| 00:00.04 | obnauticus | calling friends through it |
| 00:00.05 | obnauticus | umm |
| 00:00.08 | obnauticus | inbound lines |
| 00:01.02 | JT | this sip connection you're having a problem with, explain it |
| 00:01.12 | dlynes_laptop | JT: PCI card on a netra? |
| 00:01.21 | dlynes_laptop | JT: Does hte netra even have a pci bus? |
| 00:01.28 | JT | dlynes_laptop: it has 1 pci slot |
| 00:01.36 | obnauticus | JT i don't know how to explain it. |
| 00:01.39 | dlynes_laptop | JT: ah...never opened it up |
| 00:01.40 | JT | dlynes_laptop: i don't plan to put zap hardware in it |
| 00:01.51 | obnauticus | Like i try calling it from my cell or whatever |
| 00:01.57 | JT | obnauticus: well wtf is it? |
| 00:01.57 | obnauticus | and i can SPEAK to it through my PBX |
| 00:02.00 | dlynes_laptop | JT: I was only trying to get the software to run on it |
| 00:02.04 | obnauticus | SIP or IAX? |
| 00:02.09 | dlynes_laptop | JT: I was never planning to put it in a production environment |
| 00:02.16 | JT | yes, it's sip, i've gathered that |
| 00:02.23 | JT | but it's connecting to what? |
| 00:02.28 | obnauticus | ipkall |
| 00:02.33 | JT | dlynes_laptop: so you didn't do any testing? |
| 00:02.48 | obnauticus | insecure=very |
| 00:02.49 | obnauticus | host=voiper.ipkall.com |
| 00:02.49 | obnauticus | nat=yes |
| 00:02.49 | obnauticus | disallow=all |
| 00:02.49 | obnauticus | allow=ulaw |
| 00:02.59 | JT | ~pb |
| 00:03.12 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 00:03.12 | dlynes_laptop | JT: Not really...ran out of time |
| 00:03.12 | JT | dlynes_laptop: damn |
| 00:03.12 | dlynes_laptop | JT: Most of my time was taken up just getting the damned thing to build |
| 00:03.18 | obnauticus | i know what pastebin is. |
| 00:03.20 | obnauticus | lol. |
| 00:03.22 | JT | dlynes_laptop: i wanted to know what its capabilities were |
| 00:03.23 | obnauticus | it's not big enough |
| 00:03.31 | JT | obnauticus: then use the pastebin. |
| 00:03.35 | obnauticus | k |
| 00:03.51 | dlynes_laptop | JT: Well, fwiw, I pretty much had to transform the whole damned thing into a Linux environment |
| 00:04.09 | JT | dlynes_laptop: i'm not terribly fussed, solaris or linux |
| 00:04.17 | dlynes_laptop | JT: nobody seems to like using the Solaris build environment |
| 00:04.28 | JT | not surprised |
| 00:04.31 | dlynes_laptop | JT: so I was forced to use the GNU build environment under Solaris |
| 00:04.57 | obnauticus | JT http://papernapkin.org/pastebin/view/746 |
| 00:06.05 | JT | obnauticus: can you pastebin the whole sip.conf, minus passwords? |
| 00:06.10 | obnauticus | k |
| 00:06.19 | JT | obnauticus: also, is the call inbound or outbound? |
| 00:06.34 | JT | dlynes_laptop: hmm ok, i'm not very familiar with solaris |
| 00:06.35 | obnauticus | inbound |
| 00:06.40 | obnauticus | JT it's a default config |
| 00:06.45 | obnauticus | that's the only thing i've added |
| 00:06.51 | obnauticus | I mainly use only IAX |
| 00:06.55 | obnauticus | but for this i need to use SIP |
| 00:07.08 | JT | yeah, i have no idea what your defaults are :) |
| 00:07.17 | obnauticus | lol ok hold on |
| 00:07.27 | JT | have you checked that it's not firewalled anyway? |
| 00:07.30 | obnauticus | ya |
| 00:07.31 | *** join/#asterisk mike_jh (n=mike@85.13.255.227) |
| 00:07.32 | obnauticus | I just did |
| 00:08.02 | obnauticus | 5060 - 5082 |
| 00:08.06 | obnauticus | 4569 |
| 00:08.10 | obnauticus | 10000 - 20000 N |
| 00:08.18 | obnauticus | are all forwarded to my * server |
| 00:08.24 | JT | grr |
| 00:08.52 | JT | forwaring, i see |
| 00:08.53 | mike_jh | Does anyone know the best way to install Asterisk::AGI on Etch? It appears not to be in apt and CPAN is b0rked! Should I try fixing CPAN or is it hiddenin in apt or can I download a deb file for it? |
| 00:08.59 | obnauticus | well im using NAT |
| 00:09.04 | obnauticus | for forwarding |
| 00:09.07 | JT | i thought it was just behind nat |
| 00:09.15 | obnauticus | ya it is |
| 00:09.19 | JT | err, nat does not provide static forwarding |
| 00:09.19 | obnauticus | im using m0n0wall |
| 00:09.19 | obnauticus | ok |
| 00:09.53 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.178.29) |
| 00:10.02 | JT | does anyone actually connect directly to your asterisk machine as if it's a sip or iax server? |
| 00:10.19 | obnauticus | Clients connect as IAX |
| 00:10.25 | obnauticus | through IAX** |
| 00:10.41 | obnauticus | Do you mind if i pastebin something that is in unix formatting |
| 00:10.45 | obnauticus | like all the line breaks are fucked up |
| 00:10.51 | obnauticus | but if you have dreamweaver you can get rid of that |
| 00:11.11 | JT | wtf would i need dreamweaver to format text? |
| 00:11.11 | Nugget | wha? |
| 00:11.22 | obnauticus | well it does that for you if you just open it up with it |
| 00:11.26 | obnauticus | that's what i do |
| 00:11.26 | obnauticus | lol. |
| 00:11.30 | JT | uhuh |
| 00:12.05 | JT | i wouldn't be caught dead with dreamweaver on my pc |
| 00:12.14 | obnauticus | http://papernapkin.org/pastebin/view/748 |
| 00:12.18 | obnauticus | I never use it |
| 00:12.29 | obnauticus | I just have it because we were doing it for some gay ass assignment at school. |
| 00:12.34 | obnauticus | I use notepad for coding lol. |
| 00:12.35 | obnauticus | and nano |
| 00:13.08 | Strom_M | a gay ass assignment, eh? |
| 00:13.21 | obnauticus | a "web development" assignment using all Design mode in dreamweaver |
| 00:13.25 | obnauticus | it was so stupid. |
| 00:13.28 | Strom_M | so you were supposed to go to the bar and catalogue the kinds of gay ass? |
| 00:13.37 | Nugget | heh |
| 00:13.53 | Strom_M | the fourteen families and seventy-eight subvarieties of bubble butt? |
| 00:13.57 | rob0 | Perhaps this kid has a future in Asterisk! |
| 00:14.14 | obnauticus | huh |
| 00:14.36 | workaphobia | stoffell_w: As xkcd would put it, a gay ass-assignment |
| 00:14.44 | Strom_M | the shaved ones, the trimmed ones, the waxed ones, and the ones where the hair grows free? |
| 00:14.51 | obnauticus | JT did you see that? |
| 00:15.09 | rob0 | NM, just some gay ass jokes. |
| 00:15.36 | *** join/#asterisk nohop (n=nohup@cc501678-a.hgv1.dr.home.nl) |
| 00:15.43 | nohop | hey ppl... |
| 00:15.49 | nohop | sorry for my rudeness yesterday :) |
| 00:16.12 | rob0 | Sorry for mine, too. No hard feelings. |
| 00:16.21 | mike_jh | Oh, did I miss something exciting? |
| 00:16.32 | nohop | not really :) |
| 00:16.38 | rob0 | (Just play along with us.) |
| 00:16.41 | mike_jh | Bummer |
| 00:17.03 | nohop | i had a bad day and then someone here was yelling at me... so i parted with "fuck this" as my last message or smth :) |
| 00:17.06 | Strom_M | I can make more gay ass-jokes if you feel you need amusement |
| 00:17.07 | rob0 | I don't remember it myself; perhaps was too drunk by then. |
| 00:17.09 | workaphobia | I'm sorry I offended every race, creed, gender, and God yesterday too. |
| 00:17.39 | rob0 | workaphobia: You BLEW it. You didn't offend me. TRY HARDER. |
| 00:17.41 | workaphobia | Strom_M: No, being a loyal xkcd fan, I prefer jokes of the re-arrange the hyphen type. |
| 00:17.54 | workaphobia | rob0: I love vista. |
| 00:18.08 | workaphobia | Victory! But at what cost... |
| 00:18.13 | JT | workaphobia: /that/ you shouldn't joke about |
| 00:18.19 | obnauticus | JT |
| 00:18.21 | obnauticus | did you see the config? |
| 00:18.27 | nohop | anyways... i hope someone could help me... |
| 00:18.36 | workaphobia | JT: an artist can't live within boundaries |
| 00:18.46 | nohop | i got incomming calls directed from my asterisk server to a soft-phone on my workstation... |
| 00:19.22 | nohop | but when i call that number (from my landline) it does ring, and from the landline you can hear the other side... but not vice-versa... |
| 00:20.00 | rob0 | nohop: OS? |
| 00:20.13 | nohop | linux, both boxes |
| 00:20.42 | JT | obnauticus: yes, i've noticed you're not even registering with ipkall |
| 00:20.50 | nohop | [sipsop] |
| 00:20.50 | nohop | exten => nohup,1,Dial(SIP/nohup@192.168.10.30) |
| 00:21.00 | nohop | that's in my extensions.conf... |
| 00:21.02 | rob0 | well, I meant more specific, because I suspect firewall blockage. "service iptables stop" on RH derivatives. |
| 00:21.15 | nohop | oh, and (stupid) totally forgot to paste this one |
| 00:21.17 | obnauticus | JT well it apperentally works because i can call inbound |
| 00:21.24 | nohop | Warning: 305 192.168.10.30 "Incompatible media format: None of the audio codecs is supported" |
| 00:21.27 | JT | obnauticus: apparently you assumed wrong |
| 00:21.30 | obnauticus | like with my cell phone i can call my ipkall number and it works fine |
| 00:21.42 | JT | obnauticus: and it doesn't work, because you can't hear one side of audio |
| 00:22.00 | obnauticus | i mean the line |
| 00:22.25 | JT | eh |
| 00:22.26 | obnauticus | http://www.voip-info.org/wiki-IPKall i used that guide |
| 00:23.01 | obnauticus | it doesn't umm say anywhere to register it... |
| 00:23.43 | JT | it's just a wiki |
| 00:24.27 | *** join/#asterisk Bryce3434 (n=brice@juv34-4-82-238-91-177.fbx.proxad.net) |
| 00:24.27 | JT | look at the comments, some have problems with that config |
| 00:24.27 | obnauticus | ya |
| 00:24.27 | rob0 | I don't think you DO register with ipkall ... you give them your hostname. |
| 00:24.27 | obnauticus | ya |
| 00:24.27 | obnauticus | I do too |
| 00:24.27 | JT | but it looks like ipkall lets you put in the ip |
| 00:24.37 | JT | ipkall sounds dodgy |
| 00:25.22 | rob0 | Has been pretty reliable for me. |
| 00:25.39 | JT | shrug, i don't use free voip services |
| 00:25.43 | rob0 | at least AFAIK ... I don't have to call me :) |
| 00:25.51 | obnauticus | register => **********:****@voiper.ipkall.com |
| 00:25.54 | obnauticus | should work? |
| 00:26.01 | obnauticus | except with user:pass |
| 00:26.20 | JT | obnauticus: the quickest way to start to break down the problem would be to do packet sniffing on your router |
| 00:26.57 | rob0 | I don't register for ipkall. Registration is a way to tell the SIP server where the client is. You did that with ipkall when you signed up. |
| 00:27.11 | obnauticus | ya |
| 00:27.17 | obnauticus | because that number will only need to dial your sip proxy |
| 00:27.19 | JT | rob0: it's also a way to punch through NAT |
| 00:27.28 | JT | but he has ports forwarded |
| 00:27.33 | JT | so in theory it should work |
| 00:28.24 | rob0 | ah ... nat. I just run * on my gateway machines, so I'm spared that pain. |
| 00:28.59 | obnauticus | ok |
| 00:29.03 | obnauticus | i'll look at the packets |
| 00:29.30 | JT | well before you do that |
| 00:29.36 | JT | set verbose to 10 |
| 00:29.41 | JT | switch on sip debug |
| 00:29.44 | obnauticus | k |
| 00:29.46 | JT | and pastebin a full call |
| 00:29.47 | JT | :) |
| 00:30.53 | obnauticus | how can i log output to a file in umm |
| 00:30.58 | obnauticus | the * console |
| 00:31.24 | JT | logger.conf |
| 00:31.32 | rob0 | I run the console in screen(1) and set a high scrollback buffer. |
| 00:31.43 | *** part/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net) |
| 00:31.44 | obnauticus | can putty do that |
| 00:31.45 | obnauticus | well |
| 00:31.46 | JT | enable full log or similar |
| 00:31.47 | obnauticus | im in screen |
| 00:31.49 | obnauticus | k |
| 00:32.00 | obnauticus | well |
| 00:32.06 | obnauticus | k |
| 00:33.24 | obnauticus | k |
| 00:33.26 | obnauticus | i enabled logging |
| 00:34.20 | rob0 | Timber! |
| 00:37.13 | obnauticus | ugh |
| 00:37.16 | obnauticus | i can't get logging to work |
| 00:37.17 | obnauticus | but hold on |
| 00:37.35 | rob0 | Try a chain saw. |
| 00:37.40 | obnauticus | http://papernapkin.org/pastebin/view/749 |
| 00:39.19 | obnauticus | nm |
| 00:39.19 | obnauticus | i got it |
| 00:39.28 | JT | seriously |
| 00:39.33 | JT | try pressing enter less |
| 00:40.22 | obnauticus | http://papernapkin.org/pastebin/view/751 |
| 00:43.48 | JT | obnauticus: in sip.conf, set externip to your ip |
| 00:43.55 | obnauticus | roger |
| 00:43.57 | JT | it is already in the default |
| 00:44.04 | JT | just uncomment and set |
| 00:45.17 | obnauticus | k |
| 00:45.19 | obnauticus | im trying a call |
| 00:45.51 | obnauticus | still can't hear anything from the cell phone |
| 00:46.17 | JT | set localnet too |
| 00:46.51 | *** join/#asterisk CyBeR_GeeK (n=CyBeR_Ge@201.89.88.30) |
| 00:47.19 | obnauticus | k |
| 00:47.47 | obnauticus | SUCCESS |
| 00:47.49 | obnauticus | JT i love oyu |
| 00:48.01 | JT | err, thanks :P |
| 00:48.02 | obnauticus | seriously thanks dude |
| 00:48.10 | JT | no probs |
| 00:48.11 | obnauticus | if you were here right next to me |
| 00:48.13 | obnauticus | i would blow oyu |
| 00:48.23 | JT | i didn't need to know that |
| 00:48.28 | JT | ;) |
| 00:49.32 | obnauticus | now all i need to do is get IVR menu's uop |
| 00:49.33 | obnauticus | up* |
| 00:49.35 | obnauticus | it will be leet |
| 00:53.48 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 00:57.39 | obnauticus | JT kind a noobie question |
| 00:57.42 | obnauticus | but how do i call my voicemail |
| 00:58.02 | JT | kind of a question for |
| 00:58.04 | JT | ~thebook |
| 00:58.08 | jbot | i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:58.08 | JT | ;) |
| 00:58.09 | Err | heh, give it an extension and dial it |
| 00:58.39 | obnauticus | thanks |
| 00:59.32 | obnauticus | i think that site is down |
| 00:59.33 | obnauticus | hmm |
| 00:59.39 | obnauticus | I'll just download the O"reilly book |
| 00:59.44 | Strom_M | uh |
| 00:59.49 | Strom_M | that /is/ the oreilly book |
| 00:59.57 | obnauticus | close enough |
| 01:00.00 | obnauticus | the site isn't responding |
| 01:00.03 | obnauticus | so i assumed it was different |
| 01:06.29 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 01:06.37 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 01:09.30 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 01:09.31 | *** join/#asterisk logyati (n=paulo@201.29.18.64) |
| 01:10.30 | mosty | i'm having trouble with a tc400b transcoder card, it was working ok for about half an hour, then it stopped translating to g729. i restarted asterisk and it still doesn't seem to be able to translate, and "show transcoder" is no longer available as an asterisk console command |
| 01:11.43 | ManxPower | mosty: you need to contact Digium support on monday |
| 01:12.24 | mosty | ok, i'll stick it out with gsm today then |
| 01:16.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:17.28 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:18.18 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 01:19.11 | AllanLima | CyBeR_GeeK =) |
| 01:19.39 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 01:20.05 | CyBeR_GeeK | angler :P |
| 01:20.09 | CyBeR_GeeK | AllanLima |
| 01:23.00 | *** join/#asterisk saftsack (n=saftsack@pD9E0453C.dip.t-dialin.net) |
| 01:24.32 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 01:25.24 | flenders | anyone good with cisco phones here? |
| 01:26.05 | Strom_M | just ask your question |
| 01:26.15 | obnauticus | heh i wish i HAD a cisco phone |
| 01:26.15 | obnauticus | lol |
| 01:26.26 | JT | morning flenders |
| 01:26.37 | flenders | JT: how are you amte? |
| 01:26.39 | flenders | mate |
| 01:26.45 | JT | not bad |
| 01:27.30 | flenders | well, I got a couple of 7940 and the thing does not boot up |
| 01:27.41 | JT | fun |
| 01:27.55 | obnauticus | hmm |
| 01:28.05 | obnauticus | with cisco phones they have IAX and SIP compadibility right? |
| 01:28.05 | flenders | it got an IP and I can see it trying to grab files from the tftp server, but the screen always shows 'configuring IP' |
| 01:28.38 | flenders | and I get shit loads of these on the tftp server: |
| 01:28.39 | flenders | Client 192.168.100.195:51272 /opt/tftp/OS79XX.TXT, 1 Blocks Served |
| 01:28.39 | flenders | Client 192.168.100.195:51273 /opt/tftp/SEP001121D9A601.cnf.xml, 3 Blocks Served |
| 01:29.31 | Strom_M | do those files exist? |
| 01:29.38 | flenders | yeah |
| 01:29.54 | marv | hmm, how does the graph in figure 6 at http://tools.ietf.org/id/draft-guy-iax-03.txt work? it shows bit 0 of the first octet indicating a full frame. is that right, or is it actually the 8th bit (bit 7)? e.g., if the source call number is is all zeros in the high byte, would a full frame have that byte being 0x80 or 0x01? I'm thinking 0x80, especially after looking at it in wireshark. so does that mean the diagram is wrong, or am i just u |
| 01:29.56 | Strom_M | are they readable by the user? |
| 01:30.40 | flenders | they're 644 |
| 01:31.02 | javar | need 755 |
| 01:31.41 | obnauticus | javar can you explain to me how the cisco phones work, im considoring buying one.. |
| 01:31.46 | obnauticus | they look cool |
| 01:31.46 | obnauticus | lol. |
| 01:32.20 | javar | they do not support IAX, just SIP |
| 01:32.40 | obnauticus | and t hey get their configuration through tftp? |
| 01:33.01 | javar | sure |
| 01:33.07 | obnauticus | ,,, |
| 01:33.11 | Err | marv: your question was truncated at "i just u" - but is your question just an endian-ness question? |
| 01:33.15 | JT | obnauticus: pretty much almost no hardphones support iax |
| 01:33.43 | JT | none worth using anyway |
| 01:33.52 | obnauticus | Are the cisco ones good though |
| 01:34.06 | Err | oh, wait - you're asking about within a single byte - still, it could be a question of which way they denote the bits within an octet (LSb vs. MSb) |
| 01:34.18 | marv | or am i just understanding it wrong? |
| 01:34.24 | mosty | obnauticus, you're better off getting a good non-cisco sip phone if you want to use asterisk |
| 01:34.42 | Err | ...the TCP, UDP, and IP RFCs and such use the left-most as the MSb, if it means anything |
| 01:34.49 | marv | err: yeah i'm asking about a single byte. isn't bit 7 usually the MSB in a diagram like that? |
| 01:35.12 | marv | yeah, i'm thinking left most is MSb |
| 01:35.34 | flenders | javar: even tried 777, and it's still on the 'configuring IP', even though it has an IP. |
| 01:35.43 | marv | in which case, i'm thinking either the numbering is wrong, or i'm just not familiar with this convention |
| 01:35.46 | Err | it is in the old RFCs - I don't know, that's probably a standard somewhere for RFCs... So your packet capture data jiving with 0x80 and not 0x01 is what I would expect |
| 01:36.06 | obnauticus | mosty ok |
| 01:36.09 | obnauticus | like a linksys one? |
| 01:36.10 | obnauticus | well |
| 01:36.15 | obnauticus | that is kinda like cisco lol. |
| 01:36.40 | mosty | obnauticus, some linksys phones are ok. most people prefer polycom though |
| 01:36.46 | obnauticus | k |
| 01:36.54 | javar | flenders, you're sure that the tftp is running? |
| 01:37.01 | marv | Err: what is in old RFCs? |
| 01:37.07 | flenders | yeah, it's trying to access it |
| 01:37.09 | obnauticus | an old RFC |
| 01:37.10 | obnauticus | lol |
| 01:37.13 | obnauticus | RFC1 |
| 01:37.14 | flenders | Client 192.168.100.195:50301 /opt/tftp/OS79XX.TXT, 1 Blocks Served |
| 01:37.14 | flenders | Client 192.168.100.195:50302 /opt/tftp/SEP001121D9A601.cnf.xml, 3 Blocks Served |
| 01:37.18 | Err | marv: that's the traditional numbering style, too - it's just not exactly intuitive. They're numbered based on what order they go On The Wire, I think (for a "standard" link - probably X.25 or ethernet) |
| 01:37.28 | javar | you can access to file over browser? |
| 01:37.39 | javar | maybe your xml are wrong |
| 01:37.51 | obnauticus | mosty why is umm a cisco phone bad for asterisk... |
| 01:37.57 | obnauticus | i mean if i can get it to work on SIP |
| 01:37.58 | marv | Err: ah, ok. |
| 01:38.01 | marv | thanks |
| 01:38.08 | Err | marv: that's how 791 is written, and I just checked - the usual 0x45 for the first byte only makes sense if the bit 0 is the MSb |
| 01:38.13 | JT | because cisco are arseholes |
| 01:38.19 | flenders | javar: it could be... I got the XML files from: http://www.voip-info.org/wiki/view/cisco+mass+deployment |
| 01:38.37 | mosty | obnauticus, if you can afford a cisco service contract, which you need to get the sip firmwares, then you can afford to use cisco for the server too |
| 01:38.43 | JT | if you buy a second hand cisco, you can't even legally have the right to run the firmware that's on it, let alone get an upgrade |
| 01:38.49 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 01:39.05 | JT | and sip plays second fiddle to sccp at cisco |
| 01:39.06 | obnauticus | mosty lol. |
| 01:39.14 | obnauticus | That's so gay. |
| 01:39.14 | *** join/#asterisk swift99 (n=dave@ip70-178-9-136.ks.ks.cox.net) |
| 01:39.33 | marv | I'm used to, say, an ASCII vs UTF-8 discussion, where we talk about the 8th bit, which in that diagram would be bit 0 and not bit 7 as i would except. but i guess it's just a different conversion, that's cool. |
| 01:39.53 | JT | obnauticus: so's talking like a 15 year old :) |
| 01:40.08 | obnauticus | JT well take into considoration that I am 15 |
| 01:40.14 | JT | :P |
| 01:40.19 | swift99 | Question for gurus: Does asterisk run well on 64 bit architectures yet? |
| 01:40.21 | obnauticus | and I could be abbreviating the word "you" or "are" |
| 01:40.24 | obnauticus | but even that bothers me. |
| 01:40.28 | Err | ah, crisco - the company that can't ever remember that without standards they wouldn't have ever started |
| 01:40.32 | obnauticus | To an extent where i cannot explain. |
| 01:40.35 | javar | flenders, try sep001121d9a601.cnf.xml |
| 01:40.38 | obnauticus | im going to eat so brb :) |
| 01:40.43 | javar | change the letters |
| 01:40.57 | flenders | casing? |
| 01:41.27 | javar | yes |
| 01:41.42 | Err | marv: see rfc 990 - it describes the diagram order |
| 01:42.14 | flenders | javar: cp SEP001121D9A601.cnf.xml sep001121D9A601.cnf.xml ? |
| 01:42.18 | *** join/#asterisk nighty^^ (n=nighty@sushi.rural-networks.com) |
| 01:42.24 | javar | yes |
| 01:42.46 | Err | swift99: it runs just fine on my AMD64 box, to the extent that I use it - but I don't use most of the features of asterisk |
| 01:42.59 | marv | thanks, that makes it pretty clear |
| 01:43.49 | skymeyer | evening ;-) I want to create an inbound route on trixbox to accept incoming calls from freeworlddialup using SIP to a specific phone, but the DID number doesnt seem to be recognised |
| 01:44.01 | skymeyer | any ideas ? |
| 01:44.03 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 01:45.04 | JT | ~trixbox |
| 01:45.05 | jbot | methinks trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 01:45.14 | mosty | skymeyer, if FWD isn't sending you the dialed number info, asterisk can't tell which number was dialed |
| 01:45.17 | dudes | <skymeyer> - voip-info.org |
| 01:45.37 | flenders | javar: no luck |
| 01:46.03 | skymeyer | jbot: thanks i will do it overthere |
| 01:46.03 | jbot | skymeyer: sure thing |
| 01:46.25 | swift99 | Err: What is the scale and scope of your installation? |
| 01:47.10 | javar | flenders, In the tftp field on the phone, looks like: http://tftp.your-domain.com/spe$MA.xml |
| 01:47.30 | javar | and do just a xml file, and try |
| 01:47.30 | swift99 | Err: sorry ... geek to english translation required ... How many handsets, and what features do you use? |
| 01:48.03 | skymeyer | mosty: apart from using trixbox, DID info is this the same as the destination sip address ? |
| 01:48.47 | flenders | javar: where do I check that? |
| 01:49.08 | mosty | skymeyer, no not always |
| 01:49.11 | javar | In the browser type the IP of your phone |
| 01:50.16 | javar | with admin login and advanced view, you get Provisioning tab |
| 01:50.22 | mosty | skymeyer, i know nothing about trixbox sorry |
| 01:50.28 | javar | Profile Rule: |
| 01:50.54 | skymeyer | mosty: it's not trixbox related, i was just wondering if DID is an option in the SIP messaging or just the destination sip address |
| 01:51.50 | flenders | javar: port 80 is not responding |
| 01:52.08 | mosty | skymeyer, the dialed number and sip address can be different for a call going to a sip server, but simple sip clients tend to not support that |
| 01:52.20 | javar | try **** (4) then 110# to check the IP |
| 01:52.31 | skymeyer | mosty: owkay, thx for the info |
| 01:53.42 | flenders | javar: on the keypad? |
| 01:53.48 | javar | sure |
| 01:54.07 | flenders | nothing happens |
| 01:55.42 | javar | :( |
| 01:55.46 | javar | http://www.broadvoice.com/support_install_byod_cis79xx.html |
| 01:56.03 | javar | check that to enter to web interface of the phone |
| 01:56.34 | javar | and configure it with your tftp settings |
| 01:59.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:00.37 | flenders | javar: you think P003-08-2-00 would be fine? |
| 02:00.55 | *** join/#asterisk Metfan2007 (n=metfan@189.146.139.155) |
| 02:01.10 | AllanLima | what a minimum resources to run asterisk? |
| 02:01.17 | Metfan2007 | Hi!, somebody has installed sangoma digital cards with Asterisk??? |
| 02:01.26 | flenders | root@pbx:/opt/tftp# cat OS79XX.TXT |
| 02:01.27 | flenders | P003-08-2-00 |
| 02:01.36 | mosty | Metfan2007, yes |
| 02:01.54 | flenders | AllanLima: I run it at home on a PIII 450 with 256 of ram |
| 02:01.55 | javar | flenders, really dunno.. |
| 02:01.56 | flenders | :o) |
| 02:02.13 | AllanLima | flenders hum |
| 02:02.57 | AllanLima | this is a mimimum? |
| 02:03.23 | Metfan2007 | Ok, I have problems installing wanpipe drivers, I have two A104D cards, and it appears to finish correctly, but when I execute wanrouter start I get a lot of errors |
| 02:03.43 | Metfan2007 | something about "22 - Invalid argument" in every port |
| 02:04.10 | mosty | Metfan2007, did you run wancfg and setup the card correctly? |
| 02:04.51 | JT | AllanLima: it is not a minimum, it can run on lower |
| 02:05.00 | JT | AllanLima: depends on what it needs to do |
| 02:05.35 | AllanLima | i want for tests |
| 02:05.50 | Metfan2007 | I ran the ./Setup install, and I followed all the steps, in fact, wancfg configures my zaptel.conf and zapata.conf, and at the end it sends a "Succesful" mesage |
| 02:06.04 | Metfan2007 | so I think I did everything Ok..... |
| 02:06.33 | mosty | Metfan2007, yes but you have to run wancfg and set some specific details that it can't guess automatically, did you do that? |
| 02:07.11 | AllanLima | i want to use minimum possible |
| 02:07.50 | mosty | AllanLima, then no transcoding? |
| 02:07.54 | Metfan2007 | I did not ran manually wincfg, I saw in the config process that ./setup install run wincfg at the end |
| 02:07.57 | Metfan2007 | I'll try |
| 02:08.15 | Metfan2007 | sorry, no wincfg, wancfg :) |
| 02:08.23 | AllanLima | mosty i dont know :S |
| 02:08.55 | mosty | AllanLima, there's a page called 'asterisk dimensioning' or similar on voip-info.org google for that |
| 02:09.29 | AllanLima | mosty hum |
| 02:09.35 | AllanLima | let me see |
| 02:09.36 | AllanLima | =) |
| 02:11.30 | flenders | AllanLima: mate, it all depends on what you want to do... I didn't think you could get hardware lower than what I have at home |
| 02:11.33 | flenders | :D |
| 02:11.41 | AllanLima | mosty but in this site exists minimum resources? |
| 02:12.39 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 02:12.39 | mosty | allanlima: minimum for what level of usage? |
| 02:12.59 | Metfan2007 | mosty, all the wanpipeX.conf files were already configured... |
| 02:13.30 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 02:13.33 | AllanLima | only voip |
| 02:13.43 | AllanLima | pc to pc or pc to telephone |
| 02:13.55 | mosty | Metfan2007, with the correct settings? eg E1/T1? |
| 02:14.16 | mosty | allanlima: how many simultaneous calls? |
| 02:14.38 | AllanLima | i dont know too :S |
| 02:15.03 | Metfan2007 | mosty, I hope so, in fact I already have a digium E1 card working, and I use the same configuration... |
| 02:15.18 | mosty | allanlima: then nobody can answer your question |
| 02:15.39 | Err | swift99: sorry for the delay - I'm doing some work here... I only have a handful of clients, all IP-based, and there's never more than two clients active at once (so a total of four connections - two to the outside via a VoIP provider, and the two clients themselves) |
| 02:15.44 | mosty | Metfan2007, you need to run wancfg and tell it to configure the ports as E1 |
| 02:15.47 | flenders | AllanLima: what sort of hardware you want to run asterisk on? |
| 02:16.11 | AllanLima | opteron dual 242 3.0 8gb ram :S |
| 02:16.24 | AllanLima | but this my server for webhost |
| 02:16.28 | Metfan2007 | mosty, yes, it is done, I have all the wanpipeX.conf ready, with all the paramemters, E1, CRC, HDB3, etc... |
| 02:17.01 | mosty | Metfan2007, does wanrouter hwprobe show the card? |
| 02:17.17 | mosty | Metfan2007, perhaps you need to /etc/init.d/wanrouter start? |
| 02:17.45 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:17.48 | flenders | AllanLima: so, why do you want to know what the minimum is? |
| 02:18.29 | AllanLima | not to harm webhost |
| 02:18.30 | Metfan2007 | mosty, wanrouter hwprobe shows the two cards, 8 ports in total, everything ok, and when I run wanrouter start, there is where I see the erroro messages |
| 02:18.58 | mosty | Metfan2007, paste the output from wanrouter start at a paste site, and give us the link |
| 02:21.25 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 02:23.51 | *** join/#asterisk obnauticus (n=admin@c-71-59-162-60.hsd1.wa.comcast.net) |
| 02:27.09 | Metfan2007 | mosty, you can view the errors at http://nucleum.com.mx/extras/sangomaerror.html |
| 02:28.58 | mosty | Metfan2007, it says to check /var/log/messages - can you also paste the last 50 lines or so of that? |
| 02:31.00 | flenders | javar: I can't even unlock the thing |
| 02:33.01 | Metfan2007 | mosty, http://nucleum.com.mx/extras/varlogmessages.html |
| 02:36.27 | mosty | Metfan2007, can you do tail -f /var/log/messages > somefile & wanrouter start, then paste all of "somefile" ? |
| 02:36.47 | nohop | sounds like some ethernet->phoneline thingy ? |
| 02:36.54 | Metfan2007 | mosty, after what time? |
| 02:36.54 | nohop | shit, wrong window |
| 02:38.28 | mosty | Metfan2007, i want to see /var/log/messages from right before you do wanrouter start until it finished |
| 02:38.28 | Metfan2007 | ok |
| 02:39.19 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:42.20 | Metfan2007 | mosty, http://nucleum.com.mx/extras/somefile.txt |
| 02:42.59 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:43.44 | mosty | Metfan2007, try this http://wiki.sangoma.com/wanpipe-linux-asterisk-debugging#LineDebugging |
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| 03:06.31 | swift99 | Are there Ruby bindings for asterisk? |
| 03:07.33 | mosty | swift99, all you need is a telnet module, but there are probably some higher level libs, tried google? |
| 03:07.33 | Nugget | telnet is eeeeeeevil! |
| 03:07.40 | *** part/#asterisk dudes (n=nixtux@66-216-227-31.dhcp.stcd.mn.charter.com) |
| 03:10.02 | swift99 | I'm starting with the experts (I hope). You can get so much ... er ... sludge on google, and I have to learn a whole new field (telephony) in a couple of days. |
| 03:10.31 | swift99 | I can handle telnet, especially as a programmatic interface. |
| 03:10.33 | mosty | swift99, add site:voip-info.org to your google search string |
| 03:10.43 | swift99 | excellent! |
| 03:12.08 | swift99 | mosty: That gives me much more useful info. Thanks! |
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| 03:26.06 | swift99 | Has anyone on this list had experience with telephone over cable? |
| 03:26.20 | swift99 | That's CATV cable? |
| 03:26.42 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 03:27.35 | mosty | do you mean analogue telephone equipment over cat5? |
| 03:27.56 | JT | catv is not cat 5 |
| 03:28.02 | JT | it's 75ohm coax |
| 03:28.06 | swift99 | yep |
| 03:28.16 | mosty | then no, not i |
| 03:29.05 | swift99 | I'm talking about using the data line from the cable company as a digital telephone line. I guess it depends on whether or not Cox has installed asterisk. :o) |
| 03:29.27 | JT | it will need some form of modem basically |
| 03:30.24 | swift99 | Yes ... the broadband cable modem |
| 03:30.36 | mosty | swift99, i've done voip over a cablemodem before, it works fine, but the upload speed isn't great here so you can't squeeze too many calls simultaneously |
| 03:30.47 | JT | err then it's just voice over ip over Internet |
| 03:30.58 | JT | the fact that it's cable is almost immaterial |
| 03:31.11 | swift99 | Except that you need a way for people to call in |
| 03:31.18 | swift99 | i.e. a phone number |
| 03:31.27 | mosty | swift99, get a DID with your VOIP service |
| 03:31.57 | swift99 | mosty: please elaborate - I've only been studying this for about 4 hours |
| 03:32.14 | jwh | errm |
| 03:32.14 | mosty | read the intro page at voip-info.org |
| 03:32.15 | JT | there are lots of itsps about |
| 03:32.18 | JT | ~itsp |
| 03:32.19 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 03:32.31 | swift99 | 10-4 |
| 03:33.01 | swift99 | This industry has such an alphabet soup of acronyms. |
| 03:33.28 | JT | you think telephony is bad? try the space industry (eg. nasa) |
| 03:33.42 | swift99 | Need Another Seven Astronauts? |
| 03:33.46 | swift99 | :o) |
| 03:33.49 | JT | hah |
| 03:33.52 | [TK]D-Fender | (N)eed (A)nother (S)even (A)stronauts <--------- |
| 03:33.54 | [TK]D-Fender | :D |
| 03:33.55 | dlynes_laptop | swift99: try the Java programming industry, if you think there's too many acronyms in voip |
| 03:34.01 | [TK]D-Fender | swift99, beat me to it ;) |
| 03:34.07 | JT | well |
| 03:34.16 | swift99 | I'm a java programmer |
| 03:34.22 | swift99 | I understand |
| 03:34.25 | swift99 | :o( |
| 03:34.27 | JT | it'd be great is java programming had more abbreviations |
| 03:34.32 | JT | but it's horrible |
| 03:34.34 | [TK]D-Fender | I program under the influence of Java... does that count? ;) |
| 03:34.35 | JT | it's like |
| 03:34.37 | dlynes_laptop | JT: like it doesn't already? |
| 03:34.48 | JT | print.this.to.the.screen.yes.please |
| 03:34.54 | JT | dlynes_laptop: not inside actual programming |
| 03:35.03 | JT | lots of retarded long keywords |
| 03:35.21 | dlynes_laptop | JT: s=in.readLine() ; |
| 03:35.33 | dlynes_laptop | JT: I know...too long :) |
| 03:35.53 | JT | java is a language that was designed by a committie and not programmers |
| 03:36.15 | dlynes_laptop | JT: Yeah, but Java can be as verbose, or as terse as you want it to be |
| 03:36.31 | dlynes_laptop | JT: but it's best programmed in by a bunch of fledgling university graduates |
| 03:36.45 | dlynes_laptop | JT: they're the only ones that can appreciate a truly anal language |
| 03:37.04 | JT | anyway you want it, as long as it's slow |
| 03:37.16 | *** join/#asterisk bbryant (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 03:37.19 | JT | also, it was probably most of what they were taught at uni |
| 03:37.28 | dlynes_laptop | JT: that, too |
| 03:37.46 | swift99 | Java doesn't have to be slow. However, most java programmers aren't taught how to make it perform. They are taught that performance doesn't matter. |
| 03:38.11 | dlynes_laptop | swift99: just throw more money at the hardware...the slowness will go away |
| 03:38.12 | JT | java is pretty useless for low level tasks |
| 03:38.22 | dlynes_laptop | swift99: isn't that pretty much standard Microsoft M.O.? |
| 03:38.54 | mosty | jt: you can get around that with import and temporary variables. it's the enourmousLibraryClassNamesThatIDislike |
| 03:39.14 | swift99 | JT: Yes. java models high level constructs best, and can't handle low level tasks. Java is about my 30th programming language, give or take a few |
| 03:40.39 | swift99 | dlynes_laptop: Java can be fast. Our oracle rep was blown away when a well designed but still untuned java app outperformed their native loader tool. After tuning I tripled its speed, without getting fancier on the hardware. |
| 03:41.03 | swift99 | But, most java programmers don't understand performance. |
| 03:41.22 | swift99 | Back to the topic at hand ... |
| 03:41.30 | dlynes_laptop | swift99: yes, I know Java can be fast...JBoss is a prime example |
| 03:41.53 | swift99 | EJB's are an Edsel. |
| 03:43.16 | swift99 | So, in the telephony world, for my purposes, given 11 end points and conferencing, with 5 external lines |
| 03:43.48 | swift99 | I will need a 2 to 3 GHz processor with dual independent power supply and UPS |
| 03:44.23 | swift99 | and I need a super electrical ground that is guaranteed to have no ground loops |
| 03:44.59 | swift99 | And the standard Digium board with the appropriate FXS adapters |
| 03:45.04 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:45.13 | swift99 | That sound pretty close? |
| 03:45.48 | jql | ground loops make me sad |
| 03:45.55 | JT | well i'd look into trying to get a digital PRI service instead of 5 external lines, but i guess so |
| 03:46.12 | JT | it'd work fine on a 1 point something GHz machine too |
| 03:46.19 | JT | assuming xeon |
| 03:46.50 | swift99 | I can't guarantee better than 5 externals, yet. I'll do some research andsee what the digital options are. |
| 03:47.01 | swift99 | What about a 64bit AMD? |
| 03:47.55 | JT | yes, fine, overkill even |
| 03:48.28 | swift99 | Overkill means I don't get called at 3:00 AM |
| 03:48.32 | swift99 | :o) |
| 03:48.40 | jql | nobody wants to be called at 3am |
| 03:48.42 | JT | swift99: the smallest pri service you can usually get in the US is 8 channels, and in many metro locations is cheaper than 5 pots lines |
| 03:49.02 | JT | and there are many advantages to digital |
| 03:49.07 | JT | mainly to your sanity |
| 03:49.35 | swift99 | Yes ... we have a top notch IT team, but limited telephony experience. |
| 03:49.45 | swift99 | Digital is good for us. |
| 03:50.05 | mosty | swift99, don't use FXS ports if you can avoid it |
| 03:50.12 | mosty | pci ones, i mean |
| 03:50.20 | [TK]D-Fender | DID's over digital allow you to maximize the cost of your channels, and the ability to rig callerid valuable. Also being digital signalling you don't ahve to WAIT to receive CID either, and have other options as well |
| 03:50.35 | [TK]D-Fender | PCI FXS = ASS |
| 03:50.43 | swift99 | :o) |
| 03:50.51 | [TK]D-Fender | SIP ATA / Gateway is the way to go |
| 03:50.58 | mosty | you can get 8 channel BRI in australia, probably cheaper than PRI |
| 03:51.17 | JT | mosty: most definitely NOT cheaper than PRI :) |
| 03:51.31 | jql | australia? land of Telstra? |
| 03:51.40 | JT | and others |
| 03:51.41 | mosty | jt: really? not per-channel of course |
| 03:52.08 | JT | if you can get optus pri |
| 03:52.13 | JT | $200/10ch/mo |
| 03:52.56 | JT | it works out cheaper than 3 bri circuits (6ch) |
| 03:53.12 | JT | given optus don't charge line hunt, callerid, etc etc |
| 03:53.40 | mosty | i don't pay for these things, i just try to make whatever they throw at me work |
| 03:54.02 | swift99 | To hook up to a PRI - what hardware is required? |
| 03:54.08 | JT | a PRI card |
| 03:54.24 | JT | or external pri to sip gateway, but they're costly |
| 03:54.36 | swift99 | Is it an off the shelf Digium part? |
| 03:54.46 | JT | yes, digium and others make pri cards |
| 03:54.50 | JT | there's quite a lot around |
| 03:55.03 | swift99 | ok ... so to me it's just another network adapter |
| 03:55.14 | JT | even has the same socket |
| 03:55.25 | swift99 | that could be bad |
| 03:55.32 | JT | heh |
| 03:55.34 | swift99 | but it is manageable |
| 03:55.50 | mosty | swift99, sangoma make better hardware than digium |
| 03:55.53 | swift99 | Lots of red stickers |
| 03:56.11 | swift99 | sangoma ... what's the price difference? |
| 03:57.40 | JT | the lucky thing is that the pris don't run at -48vdc like POTS |
| 03:57.40 | swift99 | :o) |
| 03:57.40 | mosty | sangoma's cost a bit more i think, but well worth it for business use |
| 03:57.40 | jql | but I like -48dc. tastes like copper |
| 03:57.41 | JT | heh |
| 03:57.41 | JT | sangoma costs less |
| 03:57.50 | [TK]D-Fender | Roughly identical cost. |
| 03:58.07 | JT | only fractionally less, but still is less |
| 03:58.48 | [TK]D-Fender | JT : I wouldn't care if it were MORE :) |
| 03:58.56 | swift99 | So, is the Sangoma A101 an example of what I would be looking for? |
| 03:59.04 | [TK]D-Fender | swift99, A101d <--------- |
| 03:59.27 | [TK]D-Fender | swift99, Yes, you absolutely want the hardware echo cancellation. |
| 04:01.16 | swift99 | So, for budget purposes I can consider it $1,000 (list) for the external PRI connector E1/T1 |
| 04:01.40 | [TK]D-Fender | swift99, a litte less |
| 04:02.10 | [TK]D-Fender | swift99, http://www.telephonydepot.com/product_p/105-052-101d.htm |
| 04:02.11 | swift99 | yes, after shopping and negotiations. It's 999.99 on voipstore.com |
| 04:02.21 | [TK]D-Fender | 890$ <- better |
| 04:02.43 | swift99 | much better |
| 04:03.32 | [TK]D-Fender | <- Miserly one |
| 04:03.42 | swift99 | What is the difference between the D and the DX? |
| 04:04.08 | swift99 | I'm the ultimate miser - I allow my self $20.00 per month for eating out. |
| 04:04.10 | [TK]D-Fender | swift99, "d" = hardware EC, "x" = PCI-X |
| 04:04.20 | swift99 | ok |
| 04:04.23 | [TK]D-Fender | swift99, No, thats just not knowing how to LIVE. |
| 04:04.31 | swift99 | it's coming together now. |
| 04:04.50 | [TK]D-Fender | swift99, there is a difference between depriving yourself of something versus paying the best price to get what you want. |
| 04:05.09 | swift99 | I live real good ... no mortgate in a couple of years, then cash for Disney vacations and computer hobbies. :o) |
| 04:05.19 | swift99 | and my crayfish |
| 04:05.34 | [TK]D-Fender | swift99, then its not what you spend on eating out thats important, its what you ARE spedning it on. |
| 04:05.57 | swift99 | I'm saving it so I can spend it on fun stuff |
| 04:06.09 | mosty | actually i think the -X in the sangoma models means PCI-express, not PCI-X (64bit PCI) |
| 04:07.23 | JT | the X in sangoma cards means pci-e, yes |
| 04:07.44 | [TK]D-Fender | Bad aim... |
| 04:07.45 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:07.51 | [TK]D-Fender | you know what they're saying... |
| 04:08.49 | [TK]D-Fender | And another advantage on Sangoma = their regular PCI models are 3.3 & 5v compliant. |
| 04:09.18 | swift99 | ok ... it makes sense now. PCI-e allows potential improvements of bus utilization of up to 4 times PCI-X, and 8 times standard PCI busses. |
| 04:09.50 | JT | what press release are you reading? |
| 04:09.55 | swift99 | So for a high utilization system I would go to more expensive PCI-e motherboard and card. |
| 04:10.01 | swift99 | Wikipedia |
| 04:10.03 | JT | no |
| 04:10.22 | JT | you would only go pci-e card because that was what's on your motherboard |
| 04:10.39 | JT | for the amount of data that's being pushed, the interface is immaterial |
| 04:10.43 | swift99 | I haven't purchased the motherboard yet |
| 04:10.58 | JT | well |
| 04:11.05 | JT | the pci cards are cheaper.... |
| 04:11.23 | swift99 | It appears that the PCI card will do the job |
| 04:11.48 | JT | that observation appears accurate |
| 04:13.17 | swift99 | What bit rate do current generation SIP phones operate at? 100MBit? 1GBit? |
| 04:13.28 | JT | 10/100 |
| 04:14.48 | swift99 | So I can't use phone lines for my data backbone. Sigh. |
| 04:14.49 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 04:14.53 | mosty | swift99, in terms of calling bandwidth, or network link? |
| 04:14.59 | JT | eh? |
| 04:15.05 | [TK]D-Fender | Damn.... blew a spoke on my brie ride today.... |
| 04:15.18 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:15.36 | swift99 | network link - we will have to support three streaming media presentation systems. I was hoping to minimize the wiring. |
| 04:15.50 | swift99 | Three extra wires won't hurt us. |
| 04:16.06 | JT | link between what and what? |
| 04:16.38 | swift99 | Link between our server room and the rest of the building. |
| 04:17.11 | JT | i'm failing to see why there would be insufficient bandwidth on these existing links |
| 04:17.58 | swift99 | Right now, nothing is existing. We're trying to optimize our resources at for new construction. |
| 04:18.09 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 04:18.10 | swift99 | Heck, the walls aren't even all up |
| 04:18.38 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 04:18.51 | swift99 | We go in wiring after the roof goes in, but before the ceiling goes in. |
| 04:19.08 | JT | how far away is this server room? |
| 04:19.09 | [TK]D-Fender | swift99, 2 Cat5E drops per station. the way to go. |
| 04:19.18 | JT | or cat6 |
| 04:19.25 | [TK]D-Fender | Yeah sure, why not |
| 04:19.30 | swift99 | We were going for Cat 6 |
| 04:20.38 | swift99 | Cat 6 is guaranteed to 1GHz, and hypothetically modeled to 10 GHz (we'll see in a few years how realtiy meets that model) |
| 04:20.45 | swift99 | Lots of bandwidth available. Just a matter of optimizing resources. |
| 04:22.20 | [TK]D-Fender | My phones CLEARLY need fiber channel! |
| 04:22.56 | swift99 | We are considering fiber backbone because of the distance (350 feet straight line). |
| 04:23.36 | mosty | i had a customer try to talk me into installing a 4G SAN interface to a box with 4 E1 lines, they were surprised when i mentioned the bandwidth of a single E1 |
| 04:24.04 | swift99 | Ok, I'll bite. What's an E1 bandwidth? |
| 04:24.08 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 04:24.41 | mosty | about 2M |
| 04:24.43 | JT | swift99: umm |
| 04:24.50 | mosty | from memory |
| 04:24.51 | JT | swift99: NOT 10GHz |
| 04:25.16 | JT | 500MHz |
| 04:25.40 | [TK]D-Fender | 4.77mhz XT FTW! |
| 04:25.48 | JT | actually, it's only guaranteed to 250MHz |
| 04:25.57 | swift99 | JT: Sorry, that's 1GB and 10GB. You're right. It's after mybed time. |
| 04:26.21 | JT | swift99: bitrate != frequency bandwidth |
| 04:26.42 | swift99 | Yes, when I'm awake I'm aware of that. My bad. |
| 04:27.01 | JT | 10GHz would be absolutely spectacular for twisted pair, as it stands, you either need waveguide or $30/m coax to do 10GHz |
| 04:27.10 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 04:27.26 | b1shop | [TK]D-Fender: dumb question. what did i set the root passwd to? ;-P |
| 04:27.52 | JT | [TK]D-Fender: where did I leave my car keys? |
| 04:28.06 | [TK]D-Fender | JT..... I wasn't jsut working on your car ;) |
| 04:28.12 | JT | heh |
| 04:28.18 | [TK]D-Fender | JT... I WAS just working on his system. |
| 04:28.18 | swift99 | Hey, thanks all for your help. JT, mosty, [TK]D-Fender, dLynes-laptop, and waKKu |
| 04:28.25 | JT | np |
| 04:28.32 | JT | [TK]D-Fender: thought it was something like that |
| 04:28.59 | swift99 | I need my beauty sleep (ok, we'll worry about the beauty later) if I'm going to work tomorrow. |
| 04:29.15 | swift99 | cheers! |
| 04:30.22 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 04:30.40 | [TK]D-Fender | b1shop, I e-mailed it to you |
| 04:32.14 | b1shop | send in pm |
| 04:33.34 | DeL3e7 | default IAX port 5060? |
| 04:34.21 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-70-107-173-57.ny325.east.verizon.net) |
| 04:34.47 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 04:35.04 | JT | that's sip |
| 04:35.08 | JT | iax is udp 4569 |
| 04:35.28 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:37.21 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 04:43.59 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 04:45.38 | DeL3e7 | this is confusing stuff |
| 04:47.15 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 04:48.25 | k31th | is it possible to provision snom 360 phones to use active directory ? |
| 05:01.36 | obnauticus | Does * 1.4 have DigitTimeout |
| 05:01.38 | obnauticus | because umm |
| 05:01.43 | obnauticus | pbx.c:1797 pbx_extension_helper: No application 'DigitTimeout' for extension (mainmenu, s, 3) |
| 05:07.47 | [TK]D-Fender | obnauticus, taht is GONE in 1.4 |
| 05:07.59 | obnauticus | what should i use?? |
| 05:08.03 | obnauticus | Wiat? |
| 05:08.06 | obnauticus | Wait* |
| 05:08.09 | [TK]D-Fender | obnauticus, Time to read all of the upgrade.txt's that were released |
| 05:08.16 | [TK]D-Fender | obnauticus, "show function TIMEOUT" |
| 05:09.03 | bintut | hello all.. will this card/chipset work on asterisk ==> http://paste.debian.net/31292 |
| 05:10.14 | [TK]D-Fender | bintut, lol. Try it and see, and prepare for the realization that it is all likelyhood completely worthless. |
| 05:10.32 | obnauticus | What about ResponseTimeout |
| 05:11.14 | bintut | [TK]D-Fender: you mean, x100p clones are useless? :( |
| 05:11.45 | mosty | bintut, usually crappy enough to be more pain to use than buying something better |
| 05:11.45 | flenders | alright, I finally got one of my cisco 7940s to work |
| 05:11.51 | [TK]D-Fender | bintut, No, they are NEXT to worthless. Your card on the other hand may be ont he other side of the fence entirely. |
| 05:12.05 | flenders | and comparing it's sound quality to polycoms, polycoms are a lot better |
| 05:12.47 | obnauticus | [TK]D-Fender |
| 05:12.56 | obnauticus | What is the 1.4 equivilant to responseTimeout |
| 05:13.12 | [TK]D-Fender | obnauticus, I just answerd your squestion, pay attention. |
| 05:13.33 | obnauticus | That is DigitTimeout |
| 05:13.40 | [TK]D-Fender | BOTH |
| 05:13.43 | obnauticus | ohh |
| 05:13.46 | obnauticus | i didn't know that |
| 05:13.59 | [TK]D-Fender | obnauticus, If you REALLY read the INSTRUCTIONS, you' wouldn't have had to ask the 2nd one. |
| 05:14.13 | obnauticus | ya i didn't really pay attention to the Show function timeout |
| 05:14.14 | obnauticus | lol |
| 05:14.16 | obnauticus | .sorry |
| 05:14.18 | obnauticus | :( |
| 05:18.58 | obnauticus | ok [TK]D-Fender |
| 05:19.05 | obnauticus | exten => s,3,Timeout(5) |
| 05:19.11 | [TK]D-Fender | ...... |
| 05:19.12 | obnauticus | [2007-06-24 14:14:17] WARNING[6541]: pbx.c:1797 pbx_extension_helper: No application 'Timeout' for extension (mainmenu, s, 3) |
| 05:19.14 | obnauticus | work your magic |
| 05:19.14 | [TK]D-Fender | NO |
| 05:19.28 | [TK]D-Fender | obnauticus, TIMEOUT is a FUNCTIOn, not an APPLICATION. |
| 05:19.39 | obnauticus | ohh |
| 05:19.45 | [TK]D-Fender | *sigh* |
| 05:19.49 | obnauticus | dude [TK]D-Fender im an * noob and im reading the O'Reilly while im talking to you |
| 05:19.51 | obnauticus | so it's not that im lazy |
| 05:19.56 | obnauticus | i just get questions as i go |
| 05:21.07 | [TK]D-Fender | obnauticus, Poor approach. Read lost, try on your own, and after a long period of tragic failure including WIKI searches, THEN feel free to ask in here. |
| 05:21.37 | [TK]D-Fender | obnauticus, We are not your on-call hand-holding tutors or IQ. |
| 05:21.56 | obnauticus | You, however are an asterisk master. |
| 05:22.02 | [TK]D-Fender | obnauticus, If you WANT comprehensive lessons, I'm sure there a re a number of consultants who'd be up to the task however. |
| 05:22.05 | obnauticus | And are willing to help a hot man on call. |
| 05:22.55 | [TK]D-Fender | obnauticus, I am a CONSULTANT. When people want things handed to them all the way, thats business. When people who put in the effort and need only the little stuff sure I help out TONS in here. |
| 05:23.02 | *** join/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 05:23.10 | *** part/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 05:23.25 | [TK]D-Fender | obnauticus, but failing advise I've already given is a bad "X" on your record with me :) |
| 05:23.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:23.42 | obnauticus | well i just assumed that when i said umm |
| 05:23.51 | obnauticus | DigitTimeout you would give me the equivilant application |
| 05:23.59 | obnauticus | i did not know t hat one was an application and one was a function |
| 05:24.03 | obnauticus | is asking that question so bad? |
| 05:25.13 | *** join/#asterisk YonahW-Work (n=YonahW-W@genie03-173-74.inter.net.il) |
| 05:25.21 | [TK]D-Fender | <[TK]D-Fender> obnauticus, "show function TIMEOUT" <- you a) missed the giant keyword FUNCTION. and b) clearly didn't read it thoroghly otherwise you'd have realised your other query is answered in there as well. |
| 05:25.50 | obnauticus | I didn't even know there were Fucntions |
| 05:25.51 | [TK]D-Fender | I mean REALLY.. who do you think you're fooling here? :) Own up and GROW! |
| 05:25.56 | obnauticus | i thought thwy were all applicatiopns |
| 05:25.59 | [TK]D-Fender | obnauticus, Keep reading ;) |
| 05:26.02 | obnauticus | Umm...I'm 15 |
| 05:26.10 | DeL3e7 | do not question the mighty Obnauticus. because silence is his wisdom |
| 05:26.10 | obnauticus | you should be glad im not going to someone else and pming them to be spoon fed |
| 05:26.15 | [TK]D-Fender | Good... then you can't say your growth is capped yet ;) |
| 05:26.22 | obnauticus | DeL3e7 ROFL YOU'VE SEEN THAT?! |
| 05:26.45 | DeL3e7 | havent seen the movie yet |
| 05:26.49 | DeL3e7 | kinda pissed |
| 05:26.50 | obnauticus | Oh god. |
| 05:26.53 | obnauticus | It's so lame |
| 05:26.56 | DeL3e7 | didnt last in theatres long |
| 05:26.58 | obnauticus | it's EXACTLY what you would expect. |
| 05:28.05 | [TK]D-Fender | Which? |
| 05:28.17 | obnauticus | we're talking about aqua teen hunger force. |
| 05:28.20 | obnauticus | So umm... |
| 05:28.56 | [TK]D-Fender | Yay. Mooninites.... |
| 05:29.08 | [TK]D-Fender | I'm so glad I don't actually watch TV anymore... |
| 05:29.13 | DeL3e7 | plutonians |
| 05:31.01 | obnauticus | lucky ass. |
| 05:31.02 | obnauticus | ugh |
| 05:31.38 | obnauticus | and [TK]D-Fender |
| 05:31.45 | obnauticus | the reason why i came here is becasue the book was outdated |
| 05:31.46 | obnauticus | so umm |
| 05:31.55 | obnauticus | I on;y neglected to google |
| 05:32.57 | [TK]D-Fender | Luck? nothing of the sort. |
| 05:33.13 | [TK]D-Fender | intelligent purchases and a chosen style of implementation. |
| 05:33.22 | [TK]D-Fender | and I'm ditching my 52" HDTV for it |
| 05:33.37 | obnauticus | Umm |
| 05:33.38 | [TK]D-Fender | obnauticus, Functions are current and they are well listed int he book. |
| 05:33.51 | obnauticus | they have a 2000$ 1080p projector |
| 05:33.52 | obnauticus | i forogot where |
| 05:33.53 | [TK]D-Fender | they came out in 1.2 |
| 05:34.07 | obnauticus | well the book apperentally still lists the responsetimeout function |
| 05:34.11 | obnauticus | and has it's perameters and all |
| 05:34.35 | [TK]D-Fender | obnauticus, well nothing I watch requires more than 480i so my $500 SVGA projector is odoing the job just FINE . |
| 05:34.48 | obnauticus | But you MUST have 1080p |
| 05:34.49 | [TK]D-Fender | obnauticus, Well taht did still exist in 1.2 |
| 05:34.50 | obnauticus | lol. |
| 05:34.55 | obnauticus | ya |
| 05:35.08 | [TK]D-Fender | 1,.4 nuked all the deprecated stuff. |
| 05:35.14 | [TK]D-Fender | thats what reading upgrade.txt is for |
| 05:35.18 | obnauticus | i am |
| 05:35.24 | obnauticus | im looking at dialplan funcitons rightn ow |
| 05:35.31 | [TK]D-Fender | excellent |
| 05:35.53 | [TK]D-Fender | ok, well its bed-time here. best of luck with your efforts |
| 05:42.07 | *** join/#asterisk danp (i=danp@elmer.glueless.net) |
| 05:42.12 | *** join/#asterisk eliyahud (n=eliyahud@ool-182f9fe7.dyn.optonline.net) |
| 05:43.10 | danp | yo, what firmware version are polycom users using? |
| 05:43.26 | eliyahud | how do i have asterisk place a call to someone so that when they answer, they're dropped into a particular conference? |
| 05:45.00 | mosty | use a call file |
| 05:45.10 | mosty | or the originate command |
| 05:49.26 | eliyahud | oh i think I see... i use context, exten, and priority to point to an extension which drops them into that conference? |
| 05:49.36 | eliyahud | is that right? |
| 05:51.06 | mosty | yes |
| 05:52.16 | eliyahud | thanks for the help mosty |
| 05:56.57 | tzafrir | eliyahud, that's basically a dialplan setting (extensions.conf) |
| 05:57.15 | tzafrir | except: |
| 05:57.45 | tzafrir | "have asterisk make a call" can be done through a call file, manager's Originate action or a CLI Originate action |
| 05:58.01 | tzafrir | How do you need that to be initiated? |
| 05:58.45 | tzafrir | oops, haven't noticed... |
| 05:59.35 | eliyahud | well i'm setting up a web interface to our pbx |
| 05:59.50 | eliyahud | when someone clicks on a link to a particular conference room, i want them to be transferred there |
| 06:00.14 | eliyahud | this seems to be the answer |
| 06:00.24 | eliyahud | btw how do I do an originate command from the CLI |
| 06:00.42 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:01.23 | tzafrir | for a web interface, you should probably use the Asterisk Manager interface (a TCP socket, through port 5038) |
| 06:01.36 | tzafrir | and use the Originate action from there |
| 06:02.16 | tzafrir | http://voip-info.org/wiki/view/Asterisk+manager+API |
| 06:02.41 | eliyahud | yeah, actually we have already have that set up for our predictive dialer |
| 06:02.44 | tzafrir | originate from the CLI works only in 1.4 |
| 06:02.48 | eliyahud | so it looks like it shouldn't be too hard |
| 06:02.54 | eliyahud | oh thats the problem.. i'm running 1.2 |
| 06:03.08 | tzafrir | if you have 1.4, try: help originate |
| 06:03.27 | tzafrir | the idea is basically the same: original channel, and target |
| 06:03.50 | tzafrir | target is either a dialplan context, or just a dialplan line (application) |
| 06:04.52 | eliyahud | got it |
| 06:04.54 | eliyahud | toda ahi |
| 06:05.14 | tzafrir | drop by at #asterisk-il some day... |
| 06:05.29 | eliyahud | on this server? |
| 06:05.32 | tzafrir | yes |
| 06:22.30 | obnauticus | does anyone here know a solution for this |
| 06:22.31 | obnauticus | http://papernapkin.org/pastebin/view/757 |
| 06:22.39 | obnauticus | it just hangs up after the playback of sai-choose |
| 06:22.47 | obnauticus | or the background of sai-choose |
| 06:22.50 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 06:27.12 | YonahW-Work | obnauticus: maybe it is not detecting your dtmf |
| 06:27.20 | obnauticus | no it dectects it |
| 06:27.23 | obnauticus | just when the menu is done |
| 06:27.24 | obnauticus | playing |
| 06:27.27 | obnauticus | it just hangs up immidetally |
| 06:27.31 | obnauticus | even though i have a timeout set |
| 06:28.16 | YonahW-Work | sorry don't know |
| 06:28.54 | obnauticus | -- Executing [s@mainmenu:8] Set("IAX2/obnautsoft-2", "TIMEOUT(response)=15") in new stack |
| 06:28.54 | obnauticus | -- Response timeout set to 15 |
| 06:28.54 | obnauticus | == Auto fallthrough, channel 'IAX2/obnautsoft-2' status is 'UNKNOWN' |
| 06:28.54 | obnauticus | -- Hungup 'IAX2/obnautsoft-2' |
| 06:28.59 | obnauticus | that's what it says in the * console |
| 06:30.03 | YonahW-Work | well what was it supposed to do afterwards? |
| 06:30.09 | obnauticus | just wait |
| 06:30.12 | obnauticus | for a response |
| 06:30.13 | obnauticus | it's in a menu |
| 06:30.24 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
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| 06:38.39 | obnauticus | anyone know what this means: |
| 06:38.39 | obnauticus | == Auto fallthrough, channel 'IAX2/obnautsoft-1' status is 'UNKNOWN' |
| 06:40.03 | *** join/#asterisk peanutb_ (n=paulb@c-24-16-243-186.hsd1.mn.comcast.net) |
| 06:40.15 | snuffy22 | generally an unknown status can be because the peer doesn't register.. |
| 06:40.28 | YonahW-Work | obnauticus: you should check out http://www.voip-info.org/wiki/index.php?page=Asterisk%20func%20timeout |
| 06:41.11 | YonahW-Work | I think (could be wrong) you merely set what the timeout is however I don't see where you actually do anything with the timeout thus its not really of any use |
| 06:44.51 | obnauticus | well YonahW-Work im doing that |
| 06:44.52 | obnauticus | lol. |
| 06:45.15 | eliyahud | how do I get asterisk to decide which IAX or SIP trunk to dial out on.. I want to predefine limits for each trunk, and if all the concurrent channels are taken, then to dial out on the other trunk |
| 06:48.27 | YonahW-Work | obnauticus: I am pretty sure that you have to then tell asterisk to do something which it can then timeout on |
| 06:48.47 | YonahW-Work | if you want it to timeout on waiting for input while it plays the menu try setting the timeout before playing the sound |
| 06:48.57 | obnauticus | it is.. |
| 06:49.26 | kaldemar | obnauticus: you could probably use WaitExten too. |
| 06:51.37 | obnauticus | i |
| 06:51.40 | obnauticus | will check it |
| 06:51.55 | snuffy22 | eliyahud, if your provider doesn't let you make more calls than x when you dial you will get 'congested' which means u can just use another carrier |
| 06:52.32 | snuffy22 | aka if ${DIALSTATUS} = "congested do this.. |
| 06:54.51 | obnauticus | thanks kaldemar |
| 06:54.53 | obnauticus | waitexten worked |
| 06:54.59 | eliyahud | its more of a load balancing issue, I want to spread my calls evenly across several carriers |
| 06:55.05 | eliyahud | i know it sounds odd |
| 06:58.11 | kaldemar | eliyahud: read up on the dial command and voip peers and the GROUP function. that kind of function is implementable with those. |
| 07:02.38 | eliyahud | ah, this looks like exactly what I need |
| 07:02.39 | eliyahud | thanks |
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| 07:13.15 | Kadran | hello |
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| 07:18.04 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 07:18.32 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 07:24.48 | *** join/#asterisk Pilko (n=pirch@213.80.169.119) |
| 07:29.15 | *** join/#asterisk Swat2 (n=bler@218-215-199-11.people.net.au) |
| 07:29.32 | Swat2 | does asterisk support BLA (Bridged Line Appearances) ? |
| 07:29.43 | Strom_M | 1.4 does |
| 07:29.52 | Strom_M | SLA (shared line appearances) |
| 07:30.06 | Swat2 | same thing isnt it ? |
| 07:30.09 | Swat2 | sla/bla ? |
| 07:30.11 | Strom_M | yarp |
| 07:34.34 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:36.32 | *** join/#asterisk angryuser (n=aster@df01t2-212-195-112-146.d4.club-internet.fr) |
| 07:36.40 | angryuser | good day |
| 07:39.15 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:40.24 | angryuser | i have a liitle pb, when i do _4XX,1,pickup(${EXTEN:1}) when call is internal it works , when call is from external on target phone, pickup not working and i got this http://pastebin.ca/587898 |
| 07:40.31 | angryuser | he lp :) |
| 07:41.37 | Strom_M | you're not specifying the correct originating channel ID |
| 07:42.13 | Kadran | hi i try to test asterisk with ekiga softphone, and the voice quality is poor with codec G.711, any help on which codecs to use or maybe another softphone to test asterisk with |
| 07:43.25 | snuffy22 | lol g711 is the best codec in terms of voice quality |
| 07:43.26 | Strom_M | g.711 should be the best quality |
| 07:49.00 | Kadran | sure, i know but why it look so poor :( |
| 07:49.28 | Kadran | it should be so clear |
| 07:50.21 | Kadran | could anyone suggest me a softphone program that i can test asterisk with? |
| 07:50.27 | flenders | x-lite |
| 07:51.09 | purplet | hi, question: when I use the transfer feature from * (so the one in features.conf), the cid becomes the macro the extension is using... (novm/SIP{EXTEN} in my case). Anyway to change this? |
| 07:51.12 | Kadran | i have tried ekiga on windows machines. i hate windows and i think the voice is poor because of microsoft :D |
| 07:51.35 | Pilko | sjphone is nice thing too - linux or windows it works on both platforms |
| 07:51.53 | Kadran | thanks guys :) |
| 07:51.57 | JT | Kadran: the voice is poor because it's a softphones |
| 07:52.00 | JT | softphones suck |
| 07:52.51 | Pilko | about windows - you'd try switching off directx support in your client - this helped me a lot |
| 07:53.23 | Kadran | JT, i want to try it on soft phone before my manager gets me the hardphones, we still trying to pick a hardphone. |
| 07:53.48 | JT | Kadran: polycom |
| 07:54.09 | Kadran | it is my first asterisk pbx, and were thinking of snom |
| 07:54.17 | Kadran | as a hardphone |
| 07:54.22 | JT | polycom are the best |
| 07:54.31 | Kadran | thanks JT |
| 07:55.27 | flenders | Kadran: I have a linksys, a polycom, and a cisco sitting on my desk now, and they're all good. functionality on linksys is terrible, cisco is ok, and polycom is by far the best |
| 07:55.37 | flenders | sound quality and functionality |
| 08:03.46 | *** join/#asterisk angryuser (n=aster@df01t2-213-44-88-21.d4.club-internet.fr) |
| 08:03.56 | *** part/#asterisk Kadran (n=mohammed@82.201.252.155) |
| 08:04.13 | angryuser | STORM_M can you explain in details where is my error |
| 08:04.38 | *** join/#asterisk Kadran (n=mohammed@82.201.252.155) |
| 08:06.50 | angryuser | <PROTECTED> |
| 08:07.33 | angryuser | do i need to specigy what kind of extention? Sip/ZAP ? |
| 08:07.39 | angryuser | *specify |
| 08:12.42 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:19.14 | *** join/#asterisk FreezeS (n=bla@193.226.181.35) |
| 08:23.23 | *** join/#asterisk lorinc (n=ang@pool-2522.adsl.interware.hu) |
| 08:27.17 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 08:28.20 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 08:36.39 | DragoraN | how to route RTP packets in Win2k3 NAT firewall using routing and remote access |
| 08:39.17 | angryuser | <DragoraN> look for routing in wondows ;) |
| 08:39.33 | angryuser | route show&add |
| 08:39.47 | DragoraN | angryuser: NAT... :( |
| 08:39.51 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 08:39.55 | nfi|ermes | hi all |
| 08:40.05 | DragoraN | s/route/forward/g |
| 08:40.22 | angryuser | <DragoraN> you have client>>nat>>internet>>nat>> client, |
| 08:40.27 | angryuser | ? |
| 08:40.28 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 08:40.57 | DragoraN | nie :) |
| 08:41.02 | nfi|ermes | is there a asterisk log parser ?? i'd like to know the total amount of time spent on the phone, or the time spent with the numbers that begins with 055 |
| 08:41.19 | DragoraN | internet>>win2k3_router>>asterisk_with_sip>>clients |
| 08:41.23 | angryuser | <nfi|ermes> use mysql addon |
| 08:41.39 | DragoraN | my ISP provides SIP account for PSTN calls for my asterisk |
| 08:41.56 | nfi|ermes | i m going to look for it |
| 08:41.59 | nfi|ermes | thx |
| 08:42.23 | angryuser | <DragoraN> normally you need to specify externail ip in * conf and route port, that'all |
| 08:42.41 | DragoraN | angryuser, yes, but my win2k3 router is doing NAT |
| 08:42.53 | DragoraN | port 5060 forwarded, by incomding RTP packets are lost |
| 08:43.04 | angryuser | <DragoraN> mine too |
| 08:43.10 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 08:43.12 | DragoraN | :) |
| 08:43.23 | angryuser | <DragoraN> but it is not win based, but whatever nat is nat |
| 08:43.56 | DragoraN | some sip proxy i need |
| 08:47.57 | nfi|ermes | angryuser, is there a way too see if asterisk-addons is already installed ? |
| 08:48.18 | DragoraN | so how? |
| 08:48.46 | angryuser | <nfi|ermes> module show |
| 08:49.48 | angryuser | <DragoraN> http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
| 08:58.20 | *** join/#asterisk saftsack (n=saftsack@pD9E05EE2.dip.t-dialin.net) |
| 09:03.35 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 09:04.02 | *** join/#asterisk Kadran (n=mohammed@82.201.252.155) |
| 09:06.18 | *** join/#asterisk soylentgreen (n=fgast@193.238.89.34) |
| 09:07.18 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:13.24 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 09:17.03 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:17.10 | Dandre | Hello, |
| 09:17.21 | *** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru) |
| 09:17.56 | Dandre | Is there some good starting extensions.conf for 1.4 available ? |
| 09:22.03 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 09:22.05 | Kadran | hi Dandre, i use destar to configure asterisk and it looks nice to me |
| 09:22.46 | tzafrir | Dandre, extensions.conf.sample |
| 09:23.02 | tzafrir | Dandre, extensions.conf hasn't drastically changed from 1.2 |
| 09:23.55 | Dandre | but th use of users.conf seems to simplify it |
| 09:26.49 | HarryR | I wish there was a more standard method of argument parsing for dialplan commands |
| 09:26.56 | HarryR | instead of convoluted argument parsing |
| 09:27.06 | HarryR | which differs a bit from command to command |
| 09:27.37 | tzafrir | HarryR, this has improved a bit in 1.4 |
| 09:28.07 | tzafrir | Could you give an example to something that is wrong? |
| 09:28.24 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 09:29.06 | HarryR | I mean passing arguments as a hashmap of key/value arguments to the command |
| 09:29.16 | HarryR | instead of passing a string and having the command parse them it'self |
| 09:30.05 | HarryR | and have formal checkable definitions for command parameters |
| 09:31.54 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 09:33.54 | angryuser | <Dandre> make samples at the end of 1.4 install |
| 09:35.18 | *** join/#asterisk Supaplex (i=supaplex@166-70-62-199.ip.xmission.com) |
| 09:36.27 | Dandre | ok thanks |
| 09:36.31 | Supaplex | is there any way for a razr with verizon to receive forwarded voicemail, and play the attachment? I have the txt message, but the attachment was ignored. (wrong format?) |
| 09:39.16 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 09:41.04 | bintut | i'm wondering why i can't get a caller id on my pbx.. i subscribed a caller id feature from my telco and set callerid=yes on my /etc/asterisk/zapata.conf you can check the snippet of my logs at http://paste.debian.net/31308 anyone has an idea why this happens and on how to fix this? |
| 09:41.42 | Supaplex | analog line? isdn? |
| 09:42.58 | bintut | Supaplex: analog line.. i'm using the digium's tdm card |
| 09:43.14 | bintut | Supaplex: this is on my home box |
| 09:43.23 | bintut | using the digium's dev kit |
| 09:43.37 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 09:46.00 | Supaplex | does another pots device pickup the caller id ok? |
| 09:46.37 | bintut | nope |
| 09:47.35 | Supaplex | bug the phone company then. if a known working good, regular consumer device is playing dead, don't expect anything else to. |
| 09:47.51 | bintut | Supaplex: kindly check the line no. 7 of http://paste.debian.net/31308 |
| 09:48.07 | bintut | Supaplex: i don't understand it. why is that so? |
| 09:48.09 | Supaplex | I'm to tired to think |
| 09:48.10 | *** join/#asterisk beredon (n=Miranda@CPE-124-181-145-49.vic.bigpond.net.au) |
| 09:48.21 | bintut | Supaplex: it's ok.. thanks.. |
| 09:48.40 | Uatec | OMFG |
| 09:48.44 | Uatec | digium support is so crap |
| 09:48.47 | Supaplex | I'd call your telco |
| 09:48.54 | Uatec | i emailed them 4 days ago and haven't heard a word |
| 09:49.03 | Uatec | i emaled them before that about another point, never heard back. |
| 09:49.11 | Uatec | in the end i had to go to their supplier and get it sorted... |
| 09:50.27 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:51.47 | Supaplex | ffs 4am snuck up on me. again. argh. |
| 09:52.33 | negativeduck | Don't you hate it when that happens. |
| 09:55.27 | *** join/#asterisk saftsack (n=saftsack@pD9E05EE2.dip.t-dialin.net) |
| 09:57.10 | angryuser | do i need to refer the technology when i do pickup() ? sip/zap? or just extension? |
| 10:01.27 | bintut | anyone can help me here? ==> http://paste.debian.net/31311 |
| 10:01.43 | angryuser | <Uatec> digium supper helped me once |
| 10:01.47 | bintut | i don't get a caller id on my side. i'm running asterisk-1.4.5 here on my debian etch |
| 10:01.49 | angryuser | *support |
| 10:04.35 | saftsack | hi, what is the best choice for a headset for using it with a softphone? |
| 10:06.21 | angryuser | <saftsack> the are all crap more or less ;) if you dont buy a pro one buy any |
| 10:06.44 | saftsack | ok .... what is a pro one? such headsets which are delivered with snom phones? |
| 10:07.35 | angryuser | <saftsack> yea like jabra for 200€ but really i would buy a dect with fxs instead for that money |
| 10:07.58 | saftsack | thats true. so i take a really cheap one. |
| 10:08.21 | angryuser | or siemens fxs integrated dect phone |
| 10:18.00 | *** join/#asterisk aao_pwner (n=admin@c-71-59-162-60.hsd1.wa.comcast.net) |
| 10:18.48 | *** join/#asterisk Jochem (i=jochem@oosterveen.org) |
| 10:19.21 | aao_pwner | Can anyone heer explain this meetme error |
| 10:19.21 | aao_pwner | http://papernapkin.org/pastebin/view/758 |
| 10:19.31 | aao_pwner | I'm trying to join a confrence and it's umm... |
| 10:19.33 | aao_pwner | not working.. |
| 10:20.26 | tzafrir | bintut, where are you? what do you have in zapata.conf? |
| 10:20.54 | aao_pwner | tzafrir was that commenet relevant to me? |
| 10:21.12 | purplet | <aao_pwner>: do you have ztdummy loaded? |
| 10:21.32 | tzafrir | aao_pwner, no, to bintut |
| 10:21.38 | *** join/#asterisk aao_pwner (n=admin@c-71-59-162-60.hsd1.wa.comcast.net) |
| 10:21.41 | obnauticus | purplet i'm not sure |
| 10:22.01 | *** join/#asterisk aao_pwner (n=admin@c-71-59-162-60.hsd1.wa.comcast.net) |
| 10:22.04 | tzafrir | missed him |
| 10:22.10 | obnauticus | ya |
| 10:22.11 | obnauticus | lol. |
| 10:22.19 | tzafrir | aao_pwner, no, to bintut |
| 10:22.33 | obnauticus | oh ok... |
| 10:22.42 | obnauticus | purplet i don't think so |
| 10:22.58 | obnauticus | asterisk*CLI> zap show status |
| 10:22.58 | obnauticus | No Zaptel interface found. |
| 10:22.58 | obnauticus | [2007-06-24 19:18:41] WARNING[7122]: chan_zap.c:10023 zap_show_status: Unable to open /dev/zap/ctl: No such fil |
| 10:23.01 | obnauticus | If that's relevant |
| 10:24.11 | purplet | ehm.. try a "modprobe ztdummy" |
| 10:24.47 | obnauticus | erm |
| 10:24.53 | obnauticus | it gave me console |
| 10:24.58 | obnauticus | like |
| 10:25.03 | obnauticus | (/etc/asterisk)-(root@asterisk)--> modprobe ztdummy |
| 10:25.03 | obnauticus | (/etc/asterisk)-(root@asterisk)--> |
| 10:25.03 | obnauticus | ya |
| 10:25.10 | bintut | tzafrir: i'm in singapore.. my /etc/asterisk/zapata.conf is at http://paste.debian.net/31312 |
| 10:25.11 | obnauticus | j |
| 10:25.14 | obnauticus | i think it's up now |
| 10:25.17 | obnauticus | asterisk*CLI> zap show status |
| 10:25.17 | obnauticus | Description Alarms IRQ bpviol CRC4 |
| 10:25.17 | obnauticus | ZTDUMMY/1 1 UNCONFIGUR 0 0 0 |
| 10:25.44 | purplet | k, give the meetme another go then :) |
| 10:26.17 | obnauticus | thanks |
| 10:26.21 | obnauticus | how do i have it do it on startup? |
| 10:26.22 | obnauticus | lol |
| 10:27.04 | tzafrir | bintut, are you sure you have the right parameters there? (if you say so, I believe you. I honestly don't know) |
| 10:28.41 | kippi | where can I find the file that lets you set the ip asterisk listens on |
| 10:29.07 | *** join/#asterisk dec (n=tom@unaffiliated/dec) |
| 10:29.38 | dec | Hi all, how can I turn on/use directed call pickup? is **ext the right syntax? |
| 10:33.12 | *** join/#asterisk yassaccan (n=yassacca@admin212.hgo.se) |
| 10:33.17 | bintut | tzafrir: what do you mean? |
| 10:33.49 | bintut | tzafrir: i think so, as far as i understand it. |
| 10:34.18 | bintut | do i need the ztdummy driver? |
| 10:34.20 | *** join/#asterisk harryr (n=Administ@host-83-146-53-46.bulldogdsl.com) |
| 10:36.08 | tzafrir | no, you don't need ztdummy |
| 10:36.40 | tzafrir | any zaptel hardware driver should also provide timing |
| 10:36.40 | tzafrir | That is easy to test with zttest |
| 10:36.43 | obnauticus | Well umm |
| 10:39.06 | obnauticus | I juts want it to start on startup |
| 10:39.12 | obnauticus | be able to confrence |
| 10:39.35 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-87-253.red.bezeqint.net) |
| 10:41.42 | Dovid | does anyone know of a good stats program that will work with asterisk ? i need to get bandwidth stats for a box of mine |
| 10:43.35 | obnauticus | Dovid use umm |
| 10:43.36 | obnauticus | mrtg |
| 10:43.45 | obnauticus | it's not espically for asterisk.. |
| 10:43.47 | Dovid | thnx |
| 10:43.52 | bintut | tzafrir: do you have any idea why i don't get a caller id even if i already subscribed to my telco with that feature and set my /etc/asterisk/zapata.conf to usecallerid=yes ? |
| 10:43.53 | obnauticus | it monitors bandwidth usage though |
| 10:44.27 | Dovid | is there anyone that will break it down by protocol or port ? so i can see like SIP vs. IAX ? |
| 10:44.44 | obnauticus | I'm sure you could break it down to even socket... |
| 10:44.51 | obnauticus | I dunno how to do SIP vs. IAX |
| 10:45.14 | Dovid | but I can break it down based on port ? |
| 10:45.24 | obnauticus | dunno |
| 10:45.26 | obnauticus | google it :) |
| 10:45.52 | harryr | Or you could just use netflow at the router/switch side of things :) |
| 10:46.21 | obnauticus | if you got a cisco |
| 10:46.22 | obnauticus | :) |
| 10:46.28 | obnauticus | Anyone |
| 10:46.38 | obnauticus | does anyone here know of where the configuration for applications like |
| 10:46.43 | obnauticus | when you join a meetme confrence |
| 10:46.46 | obnauticus | it plays conf-hasjoin |
| 10:46.51 | obnauticus | where's the config for that |
| 10:47.06 | Dovid | config as in source ? |
| 10:47.21 | obnauticus | i gues |
| 10:47.25 | obnauticus | there has to be a config for that |
| 10:47.38 | Dovid | thats prob in the source |
| 10:48.10 | HarryR | you could just overwrite the conf-hasjoin file with your own audio? :) |
| 10:48.35 | obnauticus | well |
| 10:48.40 | obnauticus | i wanna reconfigure how long they have to say their name |
| 10:48.49 | obnauticus | i don;'t want people joining with 5min long "names" |
| 10:48.49 | obnauticus | lol |
| 10:49.47 | HarryR | "Enter your serial number, human" |
| 10:49.50 | Dovid | app_meetme.c |
| 10:49.55 | HarryR | :) |
| 10:50.00 | Dovid | seems to to be the code for it |
| 10:50.23 | obnauticus | ugh |
| 10:50.28 | obnauticus | i gotta recompile? |
| 10:52.12 | obnauticus | umm |
| 10:52.19 | obnauticus | how do i compile/install asterisk applications? |
| 10:56.10 | HarryR | see: configure & make |
| 10:57.58 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 10:59.27 | obnauticus | Do I need to do all that again |
| 10:59.33 | obnauticus | or can i just re-build the .so by it's self |
| 10:59.53 | obnauticus | and use the same compiling environment for all other applcations i may need to dev or install in the future |
| 10:59.57 | obnauticus | instead of recompiling the whole thing |
| 11:02.14 | sergee | obnauticus: config for what? |
| 11:02.22 | obnauticus | app_meetme.c |
| 11:02.29 | obnauticus | i just want to compile and link that into a .so |
| 11:02.33 | obnauticus | without recompiling all of * |
| 11:02.42 | obnauticus | like you can do with apache with umm |
| 11:02.49 | obnauticus | i forgot what it's called |
| 11:02.55 | Uatec | hey, how can i reload the voicemail.conf without restarting asterisk? |
| 11:03.07 | obnauticus | ya..type reload in the console Uatec |
| 11:03.13 | sergee | obnauticus: well, if you change only app.meeteme.c it won't recompile whole asterisk, gnu make is clever enough :) |
| 11:03.19 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:03.41 | Uatec | obnauticus, i was thinking of JUST voicemail, but hey, that works too :) |
| 11:03.48 | obnauticus | lol |
| 11:04.01 | obnauticus | well every time a voicemail box is called it re-prases that file... |
| 11:04.03 | obnauticus | i think... |
| 11:04.10 | obnauticus | it does that with extensions |
| 11:04.30 | *** join/#asterisk frawd (n=francois@89.130.32.92) |
| 11:04.32 | Uatec | ah, right :) |
| 11:04.43 | Uatec | thanks |
| 11:05.05 | sergee | obnauticus: as far as i remember timeelimit for name recording is hardcoded in app_meetme.c, you can reduce it there... |
| 11:05.20 | obnauticus | ya |
| 11:05.23 | obnauticus | I'm looking for it |
| 11:05.28 | awk | anyone here using bristuff from within debian packages? |
| 11:05.30 | sergee | obnauticus: btw, do you have any noise/crack after listening "conf-hasjoin" ? |
| 11:06.16 | obnauticus | dunno |
| 11:06.20 | obnauticus | join mine and i'll see |
| 11:06.25 | obnauticus | 3609681244 exten # at first menu |
| 11:06.27 | frawd | hello, i'm sometimes having some strange "bips" like some type of morse code while in a bridged call between SIP and analog ZAP line (TDM400P, FXO module). Anyone knows what it could be? |
| 11:07.19 | obnauticus | sergee you joining? |
| 11:08.43 | *** join/#asterisk friedrich| (n=friedric@e177250159.adsl.alicedsl.de) |
| 11:10.22 | Uatec | in sip.conf is it possible to set 2 mailboxes for a phone? |
| 11:10.29 | frawd | i have a custom kernel 2.6.18 (based on debian's one), no shared interrupts for wctdm, but sometimes zttest gives me low values (89.465332%, 98.168945%) |
| 11:10.30 | tzafrir | awk, me. Though from our own (xorcom) Debian packages, usually |
| 11:10.37 | obnauticus | Yes... |
| 11:10.44 | obnauticus | Uatec i think so |
| 11:10.52 | obnauticus | just define them with the same mailbox |
| 11:11.03 | Uatec | no, i don't mean 2 phones with the same mailbox |
| 11:11.08 | Uatec | i mean 1 phone with 2 mailboxes |
| 11:11.17 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 11:11.23 | obnauticus | umm |
| 11:11.25 | obnauticus | i dunno about that |
| 11:11.49 | sergee | obnauticus: ahh, sorry, i was afk... |
| 11:11.55 | tzafrir | frawd, 89% is really bad. Is that some sort of virtual machine? |
| 11:11.55 | obnauticus | sergee you joining?!@?!@?!@!2 |
| 11:11.55 | obnauticus | lol. |
| 11:12.09 | obnauticus | 3609681244 exten # at first menu |
| 11:13.26 | sergee | obnauticus: your number can't be reached.. |
| 11:13.29 | tzafrir | frawd, maybe high disk activity at the time? (with no dma) or other reason for excessive kernelspace activity? |
| 11:14.45 | obnauticus | are you set as blocked sergee |
| 11:14.47 | obnauticus | like |
| 11:14.53 | obnauticus | Unknown <Unknown> |
| 11:14.58 | frawd | tzafrir: it's no virtual machine |
| 11:14.58 | obnauticus | it won't accept you if so |
| 11:15.22 | frawd | tzafrir: iostats don't give me really huge activity |
| 11:15.29 | sergee | obnauticus: i dunno, probably my voipprovider can't connect to you |
| 11:15.35 | frawd | (i have SATA disks) |
| 11:15.39 | obnauticus | weird |
| 11:15.43 | obnauticus | i dunno then sergee |
| 11:15.55 | frawd | and network activity is kind of low |
| 11:16.18 | sergee | obnauticus: did you try to use meetme with several users? |
| 11:16.24 | obnauticus | Ya |
| 11:16.26 | obnauticus | three |
| 11:16.27 | obnauticus | it worked fine |
| 11:16.42 | tzafrir | vmstat / top should actually provide the data I asked regarding kernel CPU activity |
| 11:17.00 | tzafrir | anyway, do you have any other indication that the card doesn't work well? |
| 11:17.11 | tzafrir | also: is /proc/zaptel/1 the card? |
| 11:17.23 | sergee | obnauticus: can you confirm this bug? http://bugs.digium.com/view.php?id=9430 |
| 11:17.39 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 11:17.45 | obnauticus | sergee not sure |
| 11:17.47 | obnauticus | i'ld have to test it |
| 11:18.47 | sergee | obnauticus: if you will have some free time, please look at it, ok? |
| 11:18.57 | obnauticus | probably later today |
| 11:19.00 | obnauticus | i have to get some sleep first |
| 11:19.00 | obnauticus | lol. |
| 11:19.35 | sergee | obnauticus: seems like you didn't experienced this bug, and it is very interesting, can you describe your hardware? |
| 11:20.34 | tzafrir | ~pb |
| 11:20.34 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 11:20.38 | obnauticus | Well I didn't listen for it sergee |
| 11:20.38 | obnauticus | lol. |
| 11:20.43 | obnauticus | So i rally don't know |
| 11:21.06 | obnauticus | I'll be back on here later today |
| 11:21.09 | obnauticus | but right now i have to goto bed |
| 11:21.12 | obnauticus | im getting uber tired. |
| 11:21.15 | sergee | obnauticus: ok, have a good night :) |
| 11:22.07 | awk | hmm, any idea why I get this Jun 25 13:19:47 NOTICE[31987]: codec_zap.c:856 find_transcoders: No Zaptel transcoder support! |
| 11:22.20 | sergee | obnauticus: keep in mind that your input is very important, so if you'll have some free time please test meetme, and describe your experience in bugtracker, i will be very thankfull to you :) |
| 11:22.28 | awk | lsmod lists this zaptel 178212 5 vzaphfc |
| 11:23.15 | sergee | HarryR: what is "bicom"? :) |
| 11:23.23 | HarryR | ah, mistaken identity |
| 11:24.09 | awk | what is the best time to come back for somebody with information to my problem? |
| 11:24.32 | tzafrir | awk, ignore that |
| 11:24.45 | sergee | HarryR: :) |
| 11:25.00 | tzafrir | awk: ignore the zttranscode warning, that is |
| 11:25.05 | awk | tzafrir: ahh, I thought it wasn't reading my zap interface correctly and a permission issue with some nodes in /dev |
| 11:25.39 | tzafrir | zttranscode is for codec_zap.so and the Digium transcoder card |
| 11:26.07 | awk | ahh, I see.. thanks.. you wouldn't by any chance have used asterisk-bristuff from within debian at all? |
| 11:26.49 | awk | why i'm asking is that debian in their stable release has version 1.2.16 and i'm wondering if it has been patched for that BoF and DoS vulnerability or is open to exploitation.. |
| 11:27.26 | awk | I know that bristuff-1.2.18 has been released. I don't mind manually installing the package if it hasn't been patched yet |
| 11:29.44 | *** join/#asterisk kombi_ (n=kombi@213.160.14.18) |
| 11:35.01 | kombi_ | when you do make progdocs, where do those docs actually go? How do you get to see them? |
| 11:36.01 | *** join/#asterisk saftsack (n=saftsack@pD9E05EE2.dip.t-dialin.net) |
| 11:37.17 | kombi_ | a mistery, isn't it.. |
| 11:41.24 | Scrumps | probably under $prefix/share/doc |
| 11:41.46 | kombi_ | that's where I looked.. |
| 11:41.51 | Scrumps | haven't compiled the product manually yet, but that would be the logical place |
| 11:41.59 | Scrumps | ah :) |
| 11:42.06 | *** join/#asterisk Carkus (i=Zenith@cark.us) |
| 11:42.18 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 11:42.20 | Err | you would likely have to run 'make install' after building the documentation for it to show up outside of the build tree |
| 11:42.47 | Err | (I don't know about asterisk in particular, but in general non-install targets don't write outside of the current directory) |
| 11:42.59 | cy303 | sup |
| 11:43.14 | awk | hrm, asterisk doesn't come with make deinstall directives? |
| 11:43.21 | *** join/#asterisk Renacor (n=Renacor@dsl51B6D035.pool.t-online.hu) |
| 11:43.24 | awk | would be nice to add that to a future release |
| 11:43.25 | cy303 | make uninstall ? |
| 11:43.27 | Renacor | hi |
| 11:43.35 | Renacor | anybody here use sangoma cards? |
| 11:43.35 | awk | uninstall doesn't work eitehr |
| 11:43.39 | cy303 | suck |
| 11:43.43 | Scrumps | kombi_: from configure: --mandir=DIR man documentation [DATAROOTDIR/man] |
| 11:43.48 | awk | Renacor yes |
| 11:44.26 | Renacor | awk, I got this sangoma card, which is supposed to be analog, however a normal rj-11 jack does not fit into it, is there something Im missing? |
| 11:44.57 | Renacor | it's an A-200r |
| 11:45.16 | cpm | Umm, it should be an rj-45, not rj-11 |
| 11:46.06 | Renacor | umm no, for analog cards it should be rj-11 |
| 11:46.13 | Err | no, according to the website it's actually the narrow RJ-11s (like phone handsets use) |
| 11:46.18 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:46.31 | Renacor | yeah this plug is even narrower than a normal rj-11 plug |
| 11:46.32 | Err | of course, if you bother to read the website it *also* says that it comes with cables that adapt to the standard RJ-11 jacks |
| 11:47.11 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 11:47.15 | awk | Renacor as Err said, it comes with a cable, don't ask silly questions if you havent taken the time to research yourself.. |
| 11:47.17 | awk | we all busy |
| 11:48.14 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:48.31 | Err | now now, you were just asking about what patches debian adds to their packages - which is something you could've looked up too... |
| 11:49.06 | awk | Err: well fropm what I looked up they never patched it |
| 11:49.23 | Err | Renacor: if you got the card second-hand, and don't have the cables, they're a standard connector - you'll need another crimp tool, I suspect, but they're available. It's just the RJ-11 jack with 4 lines instead of 6 in it. |
| 11:53.10 | tzafrir | the docs aren't installed. Unless this is a Debian package, which is then under /usr/shared/doc/asterisk-doc/ ... |
| 11:53.21 | Renacor | jeez sorry that I offended you awk |
| 11:55.29 | Uatec | i have a group of 5 phones, when i dial a number, i want 3 of them to ring... |
| 11:55.53 | Uatec | but if one or more of the 3 are busy, or unavailable, i want the 4th and 5th to ring too... |
| 11:55.56 | Uatec | how could ido that ? |
| 11:58.04 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-175-183.buff.east.verizon.net) |
| 11:58.51 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 12:00.59 | Renacor | whats a good digium 4 port E1 card, any recommendations? |
| 12:01.20 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:02.24 | Dovid | is there any way to have asterisk not show me just one error ? one of our providers has VAD enabled and they wont turn it off. Asterisk keeps throwing the errors and it's a pin to trouble shoot |
| 12:04.20 | SuPrSluG | hello all |
| 12:05.14 | *** join/#asterisk sof76 (n=chatzill@U21d5.u.pppool.de) |
| 12:06.23 | SuPrSluG | what can cause one way audio on a zap channel. all the sip phones (polycom 500s) are on the same nat, so that should'nt cause problems. |
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| 12:28.19 | *** join/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 12:28.35 | drfreeze | Anyone here using skype as their voip service? |
| 12:28.59 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 12:33.13 | SuPrSluG | could the nat=yes setting in sip.conf cause one way audio? |
| 12:35.30 | drfreeze | SuPrSluG: yes |
| 12:35.54 | drfreeze | SuPrSluG: check that it's not the phones fault too |
| 12:36.17 | SuPrSluG | how can I check that it's the phones fault? |
| 12:36.38 | drfreeze | does the handset work? |
| 12:36.53 | drfreeze | can you test it on a working trunk? |
| 12:36.55 | SuPrSluG | or what should I look for in sip debug? |
| 12:37.08 | sof76 | Hi all, I have a problem when launching asterisk with the safe_asterisk command, if I try then asterisk -rx "stop gracefully" I get Asterisk ended with exit status 139 Asterisk exited on signal 11 Automatically restarting Asterisk, can someone help me? thanks |
| 12:37.11 | sof76 | I am running asterisk 1.4 on a virtual private server and I did not install zaptel and libpri |
| 12:37.21 | drfreeze | SuPrSluG: you can set full logs and watch the log file |
| 12:37.53 | SuPrSluG | i have that set. not getting that much rtp info. |
| 12:38.20 | drfreeze | sof76: sounds like you need to edit modules.conf and tell it to not load a missing module |
| 12:38.24 | SuPrSluG | which is where the audio stream is |
| 12:38.37 | drfreeze | SuPrSluG: post your sip.conf file |
| 12:40.16 | drfreeze | SuPrSluG: I have heard this problem due to 3 things. 1) not using friend, 2) bad phone, and 3) router doing cmp? redirects |
| 12:40.16 | sof76 | what type of module could it be? Can I find that in the error message? |
| 12:40.41 | drfreeze | sof76: just start asterisk with 'asterisk' and read the errors |
| 12:40.59 | drfreeze | sof76: the last one should tell you if some module is missing |
| 12:41.29 | drfreeze | SuPrSluG: icmp redirects. You can try turning off icmp redirects on your router |
| 12:41.52 | drfreeze | SuPrSluG: I'm assuming this problem has existed from the beginning and you're not debugging a new problem |
| 12:42.48 | *** join/#asterisk lovely2 (n=tylerj@fluoride.crm114.net) |
| 12:42.52 | lovely2 | yay |
| 12:42.58 | lovely2 | i got hints for queues working :) |
| 12:42.58 | SuPrSluG | no, it was actually working quite swimmingly. A storm fried the previous router and now this issue |
| 12:43.03 | lovely2 | on 1.2.18 |
| 12:43.24 | drfreeze | SuPrSluG: so it stopped working after putting in a new router? |
| 12:43.37 | SuPrSluG | drfreeze:http://pastebin.ca/588270 |
| 12:44.09 | sof76 | if I start asterisk with 'asterisk' I have no message at all |
| 12:44.25 | drfreeze | sof76: you mean it works, no problem? |
| 12:44.35 | SuPrSluG | yes. that's when the issue started. it's a remote office. and they don't communicate well. |
| 12:44.37 | drfreeze | sof76: but when you start with asterisk -rx, it quits? |
| 12:45.01 | sof76 | asterisk works |
| 12:45.12 | drfreeze | SuPrSluG: ok, so did the new router get setup identically? Ie, passing ports and all that stuff? |
| 12:45.31 | *** join/#asterisk Waverly360 (n=Waverly3@209.12.249.243) |
| 12:45.34 | drfreeze | sof76: maybe I didn't understand the problem. Thought * was silently quitting |
| 12:45.38 | Uatec | i have a group of 5 phones, when i dial a number, i want 3 of them to ring... |
| 12:45.42 | Uatec | but if one or more of the 3 are busy, or unavailable, i want the 4th and 5th to ring too... |
| 12:45.42 | sof76 | the problem is when i start it with 'safe_asterisk' and then stop it with asterisk -rx "stop gracefully" |
| 12:45.44 | Uatec | how could ido that ? |
| 12:45.57 | SuPrSluG | drfreeze:passing ports yes. i'm looking now at the other stuff. |
| 12:46.00 | sof76 | if I stop it with "stop now", I have no problem |
| 12:46.18 | drfreeze | Uatec: just add the phones to the list separating with '&' |
| 12:46.42 | Uatec | drfreeze, i won't want phones #4 and #5 to ring if #1, #2 and #3 are available though |
| 12:46.48 | Uatec | only if one or more of them is busy |
| 12:47.40 | sof76 | If I start with asterisk -c and then type stop gracefully, I get Segmentation fault |
| 12:47.52 | sof76 | with stop now, I have no error |
| 12:48.05 | drfreeze | sof76: ok, I see. Read too fast last time |
| 12:48.36 | drfreeze | sof76: is there a stop when conveninet? |
| 12:48.40 | drfreeze | *convenient |
| 12:48.57 | *** join/#asterisk matsk (n=mk@194.68.102.172) |
| 12:49.13 | sof76 | yes and it give Waiting for inactivity to perform halt |
| 12:49.14 | sof76 | Segmentation fault |
| 12:49.44 | drfreeze | Uatec: I'm not sure how to test if phones are busy. Our phones can take multiple calls, so theoretically, they are never busy. |
| 12:50.05 | drfreeze | sof76: icky |
| 12:50.06 | sof76 | for the moment I did not configure any extension |
| 12:50.17 | drfreeze | sof76: what version of * |
| 12:50.25 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 12:51.20 | sof76 | the last 1.4.5 |
| 12:51.59 | drfreeze | sof76: it may have nothing to do with the vps. It may be a bug in *. Can you try on one of your own boxes? |
| 12:52.14 | SuPrSluG | Uatec:two ways to skin that cat. you could create 2 ring groups or do it in the dial plan -> dial(sip/1&sip/2&sip/3 next priority dial(sip/4&sip/5 |
| 12:52.22 | drfreeze | sof76: I've gotta run. Good luck |
| 12:52.28 | sof76 | ok thks |
| 12:53.08 | drfreeze | SuPrSluG: good luck with the router |
| 12:53.33 | sof76 | sorry guys, I did make samples in asterisk and now the problem disappeared |
| 12:53.37 | sof76 | thank you |
| 12:53.54 | SuPrSluG | drfreeze:thanks it a simple linksys. looks as if all is normal |
| 12:54.30 | Uatec | drfreeze, i disabled call waiting, i want calls to go to voicemail if people are busy... |
| 12:55.58 | Uatec | SuPrSluG, that will only diall 1 and 2 and 3 THEN 4 and 5 |
| 12:56.37 | Uatec | i want it to ring: Default: 1, 2, 3 |
| 12:56.40 | SuPrSluG | Uatec:you can slice and dice it anyway you want. |
| 12:56.57 | Uatec | ... ? how though |
| 12:56.59 | Uatec | ? |
| 12:57.34 | SuPrSluG | Uatec:it will dial 1 2 &3, nothing available jumps to next priority 4&5 |
| 12:58.04 | Uatec | yes, but i would like it to dial 4 and 5 as well as 1 2 and/or 3 if one of the 3 is unavailable |
| 12:58.50 | SuPrSluG | then if they're busy/unavailable tell it where to go for voicemail or whatever you wnat |
| 12:59.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:59.33 | Uatec | if all 5 are busy, then i want it to go to voicemail, yes |
| 12:59.56 | Uatec | but if only 1 of the (1, 2, 3) are busy, then i want 4 or 5 to ring as well... |
| 13:00.17 | *** join/#asterisk CyberMad (n=cyber@222.124.103.38) |
| 13:00.23 | Uatec | might there be a way of polling the state of a sip channel before actually dialing? |
| 13:00.50 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:00.50 | CyberMad | does asterisk give output like smdr ? |
| 13:00.52 | SuPrSluG | yes, the priority will define whom to call. |
| 13:01.09 | SuPrSluG | and in what order |
| 13:01.22 | Uatec | i dont want it to call people in order, i want it to call them simultaneously |
| 13:02.49 | mocker | Uatec: Dial(SIP/100&SIP/200) ? |
| 13:03.10 | SuPrSluG | that's what the & does between sip/1&sip/2 it tells asterisk to dial them all. if nothing is available you tell it the next up in the batting order |
| 13:03.16 | mocker | Oh, SuPrSluG already said that. |
| 13:03.46 | Uatec | but i want it to ring 4 and 5 if there are still channels available, just not all of them... |
| 13:04.42 | CyberMad | what is the best free for call accounting? |
| 13:04.54 | waKKu | fwd ? |
| 13:05.26 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 13:05.55 | waKKu | Uatec maybe using 2 dial's ? |
| 13:06.07 | waKKu | exten 123,1,Dial(SIP/4,30,Ttr) |
| 13:06.17 | SuPrSluG | yes, but it's the concept of priority that's important |
| 13:06.18 | waKKu | exten 123,2,Dial(SIP/5,30,Ttr) |
| 13:06.53 | waKKu | on 1.4, iirc, asterisk dont use n+101 anymore |
| 13:07.04 | Uatec | waKKu, that will call them concurrently, i need it to be simultaneous |
| 13:07.08 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 13:07.10 | Uatec | also, i'm using business edition, which is 1.3 |
| 13:07.21 | waKKu | Uatec ?? <mocker> Uatec: Dial(SIP/100&SIP/200) ? |
| 13:08.15 | *** join/#asterisk klapzin (n=esdras@200.230.21.51) |
| 13:08.18 | Uatec | waKKu, but if sip/1 is unavailable i want it to ring sip/4 and sip/5 as well... |
| 13:08.20 | Uatec | at the same time |
| 13:08.22 | Uatec | not afterwards |
| 13:08.46 | klapzin | someone here knows atitude software ? |
| 13:09.05 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 13:09.07 | SuPrSluG | exten => 1,1,Dial(SIP/1&SIP/2&SIP/3,15) |
| 13:09.09 | SuPrSluG | exten => 1,2,Dial(SIP/4&SIP/5) |
| 13:09.57 | Uatec | *sigh* |
| 13:10.03 | Uatec | [14:11] <Uatec> at the same time |
| 13:10.03 | Uatec | [14:11] <Uatec> not afterwards |
| 13:10.41 | mocker | Uatec: If you dial SIP/100,200, and 300 w/ one dial command and one isn't available the others will still ring. |
| 13:11.03 | SuPrSluG | that will dial 1 2 & 3 simultaneouly, if busy/unavailable then it will dial 4&5 simultaneously |
| 13:11.24 | Kadran | hi, could someone suggest me a model for hardphone, |
| 13:11.38 | SuPrSluG | ~phones |
| 13:11.39 | jbot | somebody said phones was http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 13:11.46 | Kadran | thanks |
| 13:11.56 | Uatec | SuPrSluG, i want 4&6 to ring, even if some of 1, 2 & 3 are avaiable and ringing too, but only if one of them isn't... |
| 13:12.15 | Uatec | Kadran, SPA-922 FTW |
| 13:12.26 | Uatec | do not consider Aastras either |
| 13:12.53 | mocker | SuPrSluG: Crap, is there a list of those?? |
| 13:12.58 | Kadran | i was thinking of polycom |
| 13:13.04 | SuPrSluG | you'd probably have to write a script for that |
| 13:13.08 | mocker | Kadran: If you can get Polycom, do it. :)O |
| 13:13.18 | Uatec | i was worrying that that was the case |
| 13:13.27 | Uatec | since i have no idea how to even start that |
| 13:14.29 | SuPrSluG | you're getting into a lot of if/then type scenerios |
| 13:15.16 | Err | that seems like a strange use case |
| 13:15.17 | SuPrSluG | Uatec:how about agents |
| 13:15.29 | negativeduck | re UaTec's issue Can you pull the status of a single line via the ${DIALSTATUS} when using Dial(foo&foo&foo) I would imagine not as it would only indicate the end result of all of them? (not a solution just a question more so) |
| 13:16.34 | CyberMad | what is the best free for call accounting that read from SMDR? |
| 13:18.40 | *** join/#asterisk lorinc (n=ang@pool-2522.adsl.interware.hu) |
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| 13:19.30 | Uatec | agents? |
| 13:19.31 | Uatec | interesting |
| 13:19.37 | Uatec | but this sounds too complicated |
| 13:19.39 | Uatec | i got bored of it |
| 13:19.41 | Uatec | and my boss left |
| 13:20.03 | Uatec | now what i want to do is figure out how to dial digium's iaxtel numbers |
| 13:25.07 | Uatec | damn, the iaxtel signup page is farked |
| 13:25.19 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:25.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:28.44 | mocker | Uatec: Isn't it just 500 on a bare bones setup? |
| 13:32.07 | Vec | Does anyone have any idea, or can put me in the direction for setting up a system whereby if someone is engaged, it will autodial you and that person back when they are off the phone ? |
| 13:33.22 | Uatec | mocker, yeah, but that doens't take you to the bussiness support line |
| 13:36.54 | rob0 | It's their main menu ... "if you know your party's extension, dial it now ..." |
| 13:43.31 | lesouvage | Vec: check the callfile documentation |
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| 13:46.36 | *** join/#asterisk Hackbanger (n=hackbang@mail.newtention.de) |
| 13:47.01 | Hackbanger | hoi :) |
| 13:47.35 | HarryR | hi ho |
| 13:50.38 | lovely2 | Vec: you'll most likely need to customize something, look at callback, call file, |
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| 13:57.54 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:59.27 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 13:59.30 | *** join/#asterisk MindTheGap (n=iote@c9503fb4.bhz.virtua.com.br) |
| 13:59.59 | MindTheGap | codefreeze, how are you? |
| 14:02.19 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:03.14 | *** join/#asterisk el_4_jinete (n=root@mail.pulxar.com.co) |
| 14:03.20 | el_4_jinete | Hi all |
| 14:03.37 | Uatec | hey, does anyone have trouble holding a conversation with digium over the telephone? |
| 14:03.40 | el_4_jinete | Pls, help me I have a proble with a TE110p... |
| 14:03.42 | Uatec | i find that their lines are really bad |
| 14:04.28 | el_4_jinete | Zaptel recognize the card, and the channels are up, but asterisk does not show zaptel commands |
| 14:05.06 | el_4_jinete | What can I do? |
| 14:06.54 | el_4_jinete | Hello, who is in there? |
| 14:06.56 | Polis_ttt | have ju used modrobe zaptel and those commands first? |
| 14:07.38 | el_4_jinete | Yes, the modules are loaded |
| 14:08.29 | Polis_ttt | then i don't know, i'm not using te110-card myself, look at digium support forum, maby they got a solution :) |
| 14:08.54 | MindTheGap | waKKu, how r you? |
| 14:09.26 | *** join/#asterisk DEac- (n=deac@Platin.DenKn.de) |
| 14:09.28 | DEac- | moin |
| 14:09.47 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:09.57 | *** join/#asterisk disa (n=disa8@87.226.145.138) |
| 14:10.01 | disa | hi, all |
| 14:10.27 | disa | russkie est` ? |
| 14:10.38 | JT | nyet |
| 14:10.43 | disa | :) |
| 14:11.25 | Polis_ttt | disa: tylko angielsku |
| 14:11.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:12.37 | anonymouz666 | wtf |
| 14:14.24 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:14.49 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 14:14.58 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 14:15.01 | jeremy_g | Hi |
| 14:15.11 | jeremy_g | I need asterisk on a usb (live bootable) |
| 14:15.17 | jeremy_g | plz recommen |
| 14:15.18 | jeremy_g | d |
| 14:15.44 | JT | anonymouz666: ? |
| 14:15.48 | Polis_ttt | jeremy_g: sounds lika a very slow system that you will get |
| 14:15.56 | jeremy_g | really? |
| 14:16.01 | Polis_ttt | :) |
| 14:16.08 | JT | Polis_ttt: why would it be slow once it's booted? |
| 14:16.21 | jeremy_g | Polis_ttt:yeah, why should it be slow? |
| 14:16.35 | Polis_ttt | mysql from a usb-disc sounds pretty slow? |
| 14:16.50 | jeremy_g | :( |
| 14:17.35 | Polis_ttt | but i'm using asterisk for callcenter-solutions, so never used it for less than 5-10 users/per server, so maby it works if your not so many |
| 14:17.58 | *** join/#asterisk saftsack (n=saftsack@pD9E05EE2.dip.t-dialin.net) |
| 14:18.03 | *** join/#asterisk cheshair (n=fabio@host-78-12-100-200.cust-adsl.tiscali.it) |
| 14:18.07 | disa | ne mogu 4erez asterisk pozvonit` v sipnet. v sip.conf propisan friend "1111". asterisk zaregan na sipnet.ru. kak sovershit` zvonok v sipnet ? vot configi: http://paste.org.ru/?pree1y |
| 14:18.18 | cheshair | hi everybody! :-) |
| 14:18.20 | disa | JT, Polis_ttt ? |
| 14:19.03 | Polis_ttt | disa: i'm from sweden, not russia |
| 14:19.35 | Polis_ttt | disa: so don't speak russia ;) |
| 14:19.43 | jeremy_g | Polis_ttt:where in sweden? |
| 14:19.58 | Polis_ttt | jeremy_g: västervik/kalmar |
| 14:20.19 | jeremy_g | Polis_ttt:ok |
| 14:20.23 | disa | can u halp me? i need make incoming call from asterisk to SIP provider. conf files hier: http://paste.org.ru/?pree1y |
| 14:20.41 | disa | error: Auto fallthrough, channel 'SIP/1111-0874b000' status is 'CONGESTION' |
| 14:21.28 | Polis_ttt | jeremy_g: aa, du är också från sverige :) |
| 14:21.52 | jeremy_g | Polis_ttt:nej :) |
| 14:22.11 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:22.29 | Polis_ttt | jeremy_g: but you hostname tells me something ells :) |
| 14:23.44 | codefreeze | MindTheGap: I'm on vacation m-w. be back thursday... |
| 14:23.48 | *** join/#asterisk Vec (n=Vec@dsl-243-75-178.telkomadsl.co.za) |
| 14:23.59 | *** join/#asterisk anthm_mobile (n=anthm@000-335-686.area4.spcsdns.net) |
| 14:24.21 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 14:24.53 | MindTheGap | codefreeze, allright, have a nice time then... see you on thursday... |
| 14:25.43 | jeremy_g | Polis_ttt:you are smart! |
| 14:26.11 | Polis_ttt | jeremy_g: i know :) |
| 14:26.35 | *** join/#asterisk saftsack (n=saftsack@pD9E05EE2.dip.t-dialin.net) |
| 14:28.07 | MindTheGap | on http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 14:28.24 | JT | Polis_ttt: well he didn't mention running mysql |
| 14:28.35 | anonymouz666 | Polis_ttt hur mar du |
| 14:28.37 | anonymouz666 | lol |
| 14:28.50 | angryuser | <disa> privet |
| 14:29.01 | MindTheGap | there is a note on using g option on Dial() and NoCDR and ResetCDR as a workaround of CDR "dst" being set to s while dialing from a macro... |
| 14:29.17 | MindTheGap | anyone got this working? |
| 14:29.43 | MindTheGap | i mean, dialing from a macro and having dst to correctly show the destination, not the s priority? |
| 14:30.17 | angryuser | <MindTheGap> i have this problem dont know how to fix it |
| 14:30.30 | disa | angryuser: hi |
| 14:30.49 | *** join/#asterisk ispireuk (n=ispireuk@cust-200-57.dsl.versateladsl.be) |
| 14:31.01 | MindTheGap | angryuser, is that why you're angry? :P |
| 14:31.34 | Corydon76-work | "s" is not the priority, it's the extension |
| 14:31.42 | angryuser | <MindTheGap> no i am angry coz of misdn bri working half of a time crap |
| 14:31.50 | Dr-Linux | what does this mean? |
| 14:31.51 | Dr-Linux | Jun 25 07:29:39 NOTICE[1209]: chan_sip.c:6517 check_auth: stale nonce received from 'Irfan Shahid<sip:4094@pbx.i2cinc.com>' |
| 14:31.57 | MindTheGap | sorry, youŕe right Corydon... |
| 14:32.00 | JT | ah, using misdn, enough to make anyone angry ;) |
| 14:32.02 | ispireuk | I am trying to compile asterisk 1.4, but with ./configure I get this error: |
| 14:32.10 | ispireuk | checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no |
| 14:32.10 | ispireuk | configure: error: *** termcap support not foun |
| 14:32.17 | ispireuk | However termcap is installed |
| 14:32.29 | Corydon76-work | Dr-Linux: it means the host authenticated against a nonce that the server didn't send (or sent a long time ago) |
| 14:33.53 | MindTheGap | hey Corydon76-work, I remember you saying something like using a goto inside the macro to have this working, but couldnt realise how to do this, could you helpme? |
| 14:34.32 | Corydon76-work | MindTheGap: use the first s extension to Goto the extension you want |
| 14:35.13 | Dr-Linux | Corydon76-home: well, when i call from this sip soft client, i can hear outgoing ring, but i can't listen his/her voice |
| 14:35.13 | ispireuk | What should I install to fix that error? |
| 14:35.20 | Corydon76-work | so if the number is in ARG1, then Goto(${ARG1},1) |
| 14:35.38 | angryuser | <JT> i understand your pain ;) |
| 14:35.40 | Dr-Linux | Corydon76-home: the same extension works on other PC's though |
| 14:35.51 | JT | angryuser: i don't use misdn |
| 14:35.59 | el_4_jinete | Hi, again. I was activated the signalling pri_cpe and I can see the zap commands in the cli |
| 14:36.27 | el_4_jinete | But now I see the following error PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 14:36.28 | MindTheGap | Corydon76-work, and how do I do this? I got a pattern match before going insite the macro, so, it as ever changing number, and i cannot have pattern match inside the macro... |
| 14:36.39 | angryuser | JT and dont ever try to ! :)) |
| 14:36.46 | Corydon76-work | MindTheGap: why not? |
| 14:37.13 | MindTheGap | Corydon76-work, couse you told me so... :P |
| 14:37.13 | JT | angryuser: once was enough |
| 14:37.22 | Corydon76-work | MindTheGap: no, I didn't |
| 14:37.40 | Corydon76-work | MindTheGap: I told you that you cannot have a pattern match inside Goto's arguments |
| 14:38.44 | Corydon76-work | In other words, you cannot Goto(_XXX,1), but you can Goto(345) and the destination CAN be _XXX (which matches 345) |
| 14:38.49 | MindTheGap | Corydon76-work, oh yes, i remember... so basically I canjust goto X and have the patternmach and dial there, it it? |
| 14:39.28 | MindTheGap | Corydon76-work, i see... |
| 14:39.48 | MindTheGap | Corydon76-work, gonna try that... |
| 14:39.50 | Corydon76-work | Err, Goto(345,1) |
| 14:40.21 | SuPrSluG | el_4_jinete, what does zttool tell you |
| 14:40.45 | MindTheGap | Corydon76-work, but wont dst be changed to 345, and not 345XXXXX ? |
| 14:42.39 | angryuser | <JT> what do you use insted when you have let's say 50 users nad another case 150 users |
| 14:42.42 | angryuser | ? |
| 14:42.48 | angryuser | *instead |
| 14:43.06 | angryuser | <JT> i mean what harware |
| 14:43.25 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:44.22 | el_4_jinete | Zttool does not show me any irq misses. 0 |
| 14:44.30 | el_4_jinete | And No Alarms |
| 14:44.51 | Corydon76-work | MindTheGap: Yes. Mind that it's an example, and you should take the extension length to whatever you need. |
| 14:45.17 | ccesario | hi,,,, I need make one test with txfax and rxfax, I'm testing with my ATA and I don't give success.... Does possible to use txfax to rxfax (eg. exten => 10,1,txfax(/tmp/sample.tif|20) .... exten => 20,1,rxfax(/tmp/test.tif) ? |
| 14:46.31 | s0ck | um |
| 14:46.47 | Corydon76-work | ccesario: Have you read the documentation which says that TxFax and RxFax REQUIRE the use of HARDWARE? |
| 14:46.49 | s0ck | my b410p just got picked up as a nic :s |
| 14:47.30 | ccesario | Corydon76-work, sorry, thanks |
| 14:47.59 | MindTheGap | Corydon76-work, I have exten => s,1,Dial(sip/fwd/*1${MACRO_EXTEN:4}|60|r) . The ${MACRO_EXTEN:4} is a number mached outside the macro, so, how do I send it to the right location if I dont know the number yet? |
| 14:48.18 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 14:48.34 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:48.35 | MindTheGap | Corydon76-work, and I cant have pattern match insite goto? |
| 14:49.02 | Corydon76-work | MindTheGap: how are you calling the macro? |
| 14:49.04 | *** join/#asterisk lorinc (n=ang@pool-2000.adsl.interware.hu) |
| 14:49.04 | JT | angryuser: junghanns bri cards |
| 14:49.22 | *** join/#asterisk khronos (n=khronos@c-76-110-125-108.hsd1.fl.comcast.net) |
| 14:49.51 | MindTheGap | Corydon76-work, exten => _0001800.,3,Macro(fwdu0800) |
| 14:50.12 | el_4_jinete | SuPrSluG, Any answer? |
| 14:50.15 | Corydon76-work | MindTheGap: Use an argument... i.e. Macro(fwdu0800,${EXTEN}) |
| 14:50.22 | Corydon76-work | then use ARG1 |
| 14:52.20 | MindTheGap | Corydon76-work, and how do I match it insite the macro? i surelly cant have a $ARG1 => $ARG1,1,Dial(sip/fwd/$ARG1) |
| 14:53.18 | Mercestes | Is there a way to disable the CFwdAll softkey on the Cisco 7960? google isn't help me me much. |
| 14:53.27 | Corydon76-work | MindTheGap: buy some Digium hardware and get a Digium tech to do it for you |
| 14:53.46 | Corydon76-work | I'm not in the mood to spoonfeed you a config |
| 14:54.50 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-184-242.dhcp.embarqhsd.net) |
| 14:55.13 | angryuser | <JT> pity they do bri only , not for europe |
| 14:55.20 | MindTheGap | Corydon76-work, im not asking you to spoonfeed me, i just cant get it... |
| 14:55.23 | el_4_jinete | i, again. I was activated the signalling pri_cpe and I can see the zap commands |
| 14:55.23 | el_4_jinete | <PROTECTED> |
| 14:55.58 | el_4_jinete | But now I see the following error PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 14:56.31 | SuPrSluG | el_4_jinete, check these settings http://www.mail-archive.com/asterisk-users@lists.digium.com/msg124679.html |
| 14:56.43 | cheshair | am i allowed to post an asterisknow related question here? nobody in #asterisknow |
| 14:57.24 | Mercestes | cheshair, only if it is a rediculously easy question and we get to make fun of you for it. |
| 14:57.27 | JT | angryuser: what do you mean? |
| 14:58.55 | cheshair | asterisknow on 192.168.2.2, my distro with xlite on 192.168.2.1, i edited sip.conf as told in asteriskTFOT.pdf, i do login from xlite however i can't make calls |
| 14:58.57 | angryuser | <JT> they have 2 cards pri 1x port and 2x port ,all others are bri, in europe we use pri most of the time, they should build 4xTO pri card |
| 14:59.07 | el_4_jinete | Thank you SuPrSluG, I tell us later |
| 14:59.12 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 14:59.22 | angryuser | <JT> like b410p wuth with their drivers |
| 14:59.28 | SuPrSluG | el_4_jinete, hope it helps |
| 14:59.30 | cheshair | xlite just remains frozen saying "dialing" and nothing happens |
| 14:59.44 | JT | angryuser: err, bri is also widely used in europe, at least for smaller setups |
| 14:59.47 | JT | angryuser: umm |
| 14:59.52 | jeremy_g | Isn't there any asterisk live usb distro? |
| 15:00.03 | JT | angryuser: if you are using pri, you don't need misdn, so i don't see what your point is |
| 15:00.30 | cheshair | on the other hand, ekiga tells me: "security check failed" and it refuses to start the call |
| 15:01.08 | angryuser | JT i am using http://www.digium.com/en/products/hardware/b410p.php with misdn |
| 15:01.26 | JT | angryuser: so why are you telling me about pri then? it's irrelevant |
| 15:01.48 | angryuser | Note: The B410P is incompatible with North American BRI. |
| 15:01.59 | cheshair | Mercestes: isn't it enough ridiculous? :-) |
| 15:02.03 | JT | angryuser: still yet to see a point here |
| 15:02.13 | JT | angryuser: i'm not sure what you're trying to say |
| 15:03.10 | angryuser | <JT> well i suppose they use primary rate interface |
| 15:03.27 | JT | angryuser: so how does this relate to junghanns being somehow inferior? |
| 15:03.29 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 15:03.46 | *** join/#asterisk Falle (n=falle@87.ftth2.cust.cen2.sksk.se.borderlight.net) |
| 15:04.13 | s0ck | argh, can kudzu be invoked from the command line |
| 15:04.20 | angryuser | <JT> never said that , i sait it woul be good if they develep a card with 4t0 pri |
| 15:04.31 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:04.48 | JT | angryuser: 4t0, weird way to express 4 * PRI |
| 15:04.59 | JT | angryuser: because there's plenty of other quad pri cards around? |
| 15:05.10 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.65) |
| 15:05.14 | zeeesh | hi |
| 15:05.21 | angryuser | <JT> i dont see your point here |
| 15:05.42 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:05.49 | JT | angryuser: also can you stop quoting me and address me? it's less confusing :) eg. JT: |
| 15:06.05 | angryuser | <JT>: :) |
| 15:06.14 | JT | angryuser: there's digium and sangoma quad pri cards, it's not Junghanns' market |
| 15:06.22 | s0ck | angryuser: problems with a b410p? |
| 15:06.33 | Dr-Linux | what time digium support will open? |
| 15:06.33 | JT | angryuser: try this: jt <tab> |
| 15:07.00 | el_4_jinete | SuPrSluG, My settings are ok, But I have to use crc4 |
| 15:07.02 | tzafrir | JT, there's actually a Junghanns single and dual span E1card (its driver is cwain) |
| 15:07.10 | LeddyHM | does "canreinvite" have to be set at the trunk level, or can at the itnernal peer override it? |
| 15:07.11 | JT | i know |
| 15:07.17 | JT | tzafrir: the argument is wrt quad port |
| 15:07.25 | [TK]D-Fender | ("[{<JT>}]"),:; GOOD MORNING STARSHINE! |
| 15:07.33 | angryuser | s0ck: just little ones, my telco shut down the ports, so i set to ignore port's state and call out, but still sometime i got busy signal |
| 15:07.34 | JT | arrgh |
| 15:07.43 | JT | starshine is an irc user i know btw :P |
| 15:08.30 | [TK]D-Fender | LeddyHM, Both end s of the call have to support it. If either is marked as "no" then its a no-go. |
| 15:09.00 | *** join/#asterisk saftsack (n=saftsack@pD9E05EE2.dip.t-dialin.net) |
| 15:09.20 | LeddyHM | damn |
| 15:09.26 | LeddyHM | was trying to do a test |
| 15:09.27 | angryuser | s0ck, need to debug it one day |
| 15:09.33 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 15:10.16 | awannabe | hi guys...on a analog zap channel when a call comes in it should show something either via show channels or just in general debug right? |
| 15:10.35 | LeddyHM | actually that wasn't my question.. maybe I wasn't clear |
| 15:10.38 | angryuser | <awannabe> in cli |
| 15:10.43 | tzafrir | it should |
| 15:10.55 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 15:10.57 | LeddyHM | our sip trunk is set to reinvite=no, but the phone is set to reinvite=yes |
| 15:11.13 | awannabe | ok i thought so....and loopstart is most common, correct? |
| 15:11.14 | LeddyHM | our provider supports it, but I don't want to make a global change, just testing on 1 phone |
| 15:11.19 | Corydon76-work | The setting is "canreinvite" not "reinvite" |
| 15:11.21 | Dovid | TK: Are you comfortable using 1.4.X yet ? |
| 15:11.23 | tzafrir | awannabe, also: set verbose 3 in the CLI would give you some idea at what's going on |
| 15:11.24 | [TK]D-Fender | LeddyHM, Then the 2 won't reinvite to each other. |
| 15:11.35 | LeddyHM | tk: dang ok |
| 15:11.38 | [TK]D-Fender | Dovid, I use 1.4 personally, and have done it for a few others. |
| 15:11.41 | LeddyHM | thanks |
| 15:11.48 | Dovid | ok. i am ordering a new box |
| 15:11.50 | awannabe | tzafrir: ive got it at 10 and dont see jac |
| 15:11.52 | awannabe | jack* |
| 15:12.02 | Dovid | i know last time we spoke u weren;t sure about it |
| 15:12.10 | tzafrir | So no new channel is created. FXS or FXO? |
| 15:12.15 | *** part/#asterisk Kadran (n=mohammed@82.201.252.155) |
| 15:12.26 | b1shop | [TK]D-Fender: i found out why outbound calls always got busy signal |
| 15:12.37 | [TK]D-Fender | b1shop, ....and that is? |
| 15:13.08 | b1shop | the main line has FORCED 11-digit dialing. the other two can use 7-digit |
| 15:13.27 | [TK]D-Fender | b1shop, that is BAD |
| 15:13.34 | b1shop | no kidding |
| 15:13.37 | [TK]D-Fender | b1shop, bitch-slap your telco |
| 15:13.54 | [TK]D-Fender | b1shop, This may pose an issue for dialing. |
| 15:13.58 | angryuser | b1shop, i thought that I had acrappy telco ;) |
| 15:14.06 | b1shop | it's AT&T... |
| 15:14.11 | *** join/#asterisk seele_ (n=seele@dns.datawareltda.com) |
| 15:14.25 | awannabe | even if extensions.conf is not correct it will show calls in CLI right? (ive always used PRI never analog before!) |
| 15:14.36 | *** join/#asterisk anthm (n=anthm@h46088d27.area4.spcsdns.net) |
| 15:14.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:14.45 | [TK]D-Fender | awannabe : with PRI debung enabled, yes |
| 15:14.51 | [TK]D-Fender | debug* |
| 15:14.54 | rob0 | AT&T which? Some parts of AT&T come from different roots, and might not be as bad as others. |
| 15:15.00 | angryuser | awannabe, depends on debug |
| 15:15.02 | rob0 | yet |
| 15:15.05 | robl^ | AT&T == crappy telco == I am Mr. Big Telco and I make all the rules so deal with it |
| 15:15.05 | [TK]D-Fender | b1shop, What happens if you dial 11 digits on those other 2 lines? |
| 15:15.19 | *** join/#asterisk oej (n=olle@174.82-134-80.bkkb.no) |
| 15:15.47 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:15.53 | awannabe | i have debug set to 10 |
| 15:16.00 | *** join/#asterisk falz (n=falz@proxy.supranet.net) |
| 15:16.09 | b1shop | [TK]D-Fender: dials fine |
| 15:16.10 | [TK]D-Fender | awannabe, "pri debug span [port]" |
| 15:16.27 | b1shop | i have everyone dialing 11 digits not until i can fix it |
| 15:16.28 | [TK]D-Fender | b1shop, In a bit I will mod your dial macro to adjust for that. |
| 15:16.42 | b1shop | sweet. |
| 15:16.45 | awannabe | [TK]D-Fender: its not PRI, its analog ports |
| 15:16.58 | [TK]D-Fender | awannabe, ICk |
| 15:17.37 | [TK]D-Fender | b1shop, gimme sudo or PM the root and I'll do it now. |
| 15:18.01 | awannabe | in zapata.coonf can you do channel = or channel => ive seen both, not sure what is correct, wondering if thats it |
| 15:18.14 | [TK]D-Fender | awannabe, either IIRC. "zap show channels" <- |
| 15:18.24 | _E-bola | Have any statistics packages for asterisk showed up? |
| 15:18.29 | _E-bola | besides the old CDR records thing? |
| 15:19.27 | [TK]D-Fender | b1shop, done |
| 15:19.27 | falz | I've got a bizarre issue. I upgraded from 1.2 to 1.4.5 over the weekend. I use cisco 7960's to transfer between phones (not parking). When I opt to transfer, it never actually goes through, but the original two callers can still talk after the failed transfer. the logged message in CLI is "Spawn extension (default, 303, 1) exited non-zero on 'SIP/302-08219428" |
| 15:19.35 | syle | Can I have a vote, DO YOU use 1.2 or 1.4 branch? |
| 15:19.38 | syle | 1.2 |
| 15:19.38 | falz | (where 303 is the sender, 302 is the recipient) |
| 15:19.41 | b1shop | sweet. ty |
| 15:19.51 | [TK]D-Fender | b1shop, np, all part of the service |
| 15:19.59 | [TK]D-Fender | b1shop, we |
| 15:20.08 | anonymouz666 | anyone in here know something about provisioning pap2 configuration utility? for mass configuration |
| 15:20.20 | [TK]D-Fender | 'll still need to work out passing back the revised copy for my archives. |
| 15:20.29 | anonymouz666 | grandstream are crap but they offer something like that. |
| 15:20.46 | [TK]D-Fender | anonymouz666, So does snom Polycom, etc.... |
| 15:20.51 | [TK]D-Fender | anonymouz666, and..... |
| 15:20.52 | [TK]D-Fender | ~gs |
| 15:20.53 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:20.54 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 15:20.59 | danp | [TK]D-Fender: what polycom firmware version are you using? |
| 15:21.15 | falz | yet, per my issue above, transferring to queues works, just not directly to another SIP extension |
| 15:21.28 | toot | snoms work a treat for mass provisioning :) |
| 15:24.56 | awannabe | [TK]D-Fender: if the channels are setup right then they will show under zap show channels, correct, just like with a PRI? |
| 15:26.34 | *** join/#asterisk rantsh (n=chatzill@201.210.16.238) |
| 15:26.46 | *** join/#asterisk ManxPower (n=manxpowe@42.sub-75-200-20.myvzw.com) |
| 15:26.50 | rantsh | Hello everyone |
| 15:27.23 | rantsh | I come again with a transcoding question for all of you |
| 15:28.01 | rantsh | I should've said a set of questions, since I don't know where to start with this subject |
| 15:28.25 | anonymouz666 | PAP2 documentation sucks. |
| 15:28.45 | ManxPower | rantsh: there is very little to know about transcoding |
| 15:28.51 | ManxPower | anonymouz666: all docs for IP phones suck |
| 15:29.11 | anonymouz666 | You see in conf utility: FAX NSE mode. and then you go check manual: Enable the FAX NSE mode. |
| 15:29.22 | anonymouz666 | what the hell is NSE |
| 15:29.30 | Uatec | what does: -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 10.247.0.34 mean? |
| 15:29.40 | Uatec | i keep getting it from the same IP... |
| 15:29.43 | coppice | NSE == Not So Effective |
| 15:29.44 | rantsh | last time I came someone told me asterisk performed transcoding automatically and that for me to be able to perform it manually it'll take both parties to disagree on codecs, meaning to have someone or something on the middle accepting only 1 codec, right? |
| 15:29.48 | *** part/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 15:30.15 | ManxPower | "show translation recalc 15" will tell you the relative amounts of CPU each transcode type uses |
| 15:30.27 | Mercestes | cheshair, Sounds like Asterisknow did something wrong. |
| 15:30.53 | Mercestes | cheshair, Now you have to know what's *in* the config files to troubleshoot it which is why asterisknow is bad. |
| 15:31.21 | *** join/#asterisk colde (n=colde@pdpc/supporter/active/colde) |
| 15:31.22 | ManxPower | rantsh: If asterisk cannot make both legs of the call use the same codec OR if MeetMe is being used OR if Zap is used for 1 leg of the call then transcoding will happen if Asterisk can. |
| 15:31.32 | ManxPower | rantsh: what are you trying to ACCOMPLISH? |
| 15:31.41 | [TK]D-Fender | awannabe, yes |
| 15:31.54 | [TK]D-Fender | danp, 2.1.1 |
| 15:31.58 | ManxPower | Uatec: IT means you have debugging turned on |
| 15:32.11 | [TK]D-Fender | danp, And I've used ever version since 1.4.3 |
| 15:32.24 | cheshair | Mercestes, well i know what's in sip.conf, since i edited it by hand. i edited it accordingly to asteriskTOFT |
| 15:32.27 | awannabe | [TK]D-Fender: i got it..i had lots of typos! |
| 15:32.29 | rantsh | I want my phones to use iLBC, and my gateways use g729, I already have an * working as a b2bua in the middle |
| 15:32.36 | awannabe | thanks guys, im a just a iddiot is all! :) |
| 15:33.03 | cheshair | Mercestes, maybe asteriskNOW is not so "well-compatible" with the guide i'm reading? |
| 15:33.11 | Mercestes | cheshair, I would suggest pastebining your sip.conf, extensions.conf, and your failed call attempt text from the CLI |
| 15:33.36 | ManxPower | rantsh: So disallow=all and allow=ilibc in the sip.conf phone entry and disallow=all and allow=g729 in the gateway sip.conf entry. Make SURE you purchase G729 licenses from Digium or it won't work |
| 15:33.59 | ManxPower | rantsh: this is something people do ALL THE TIME |
| 15:34.01 | rantsh | we have the licenses already |
| 15:34.15 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 15:34.20 | ManxPower | in fact, this is the preferred way |
| 15:34.24 | rantsh | I know, I've been using asterisk for maybe 1 week now |
| 15:34.25 | ManxPower | rantsh: so what is the problem? |
| 15:35.04 | [TK]D-Fender | rantsh, transcoding is automatic. If the 2 sides can't agree on a codec for passthrough * will automatically transcode if it is able |
| 15:35.05 | ManxPower | cheshair: We don't know how to use AsteriskNOW. Ask on the channel #asterisknow |
| 15:35.05 | rantsh | My problem is I'm a n00b, and I got an assigment in something I don't know (but really want to learn) |
| 15:35.43 | rantsh | I didn't know where to start, that's all, since the docs weren't of any real help |
| 15:35.52 | rantsh | thank you though |
| 15:36.03 | [TK]D-Fender | cheshair, AsteriskGUI uses "USERS.CONF" for its config and it will OVERRIDE sip.conf zaptel.conf and other such things. users.conf is a flaming pile of SHIT IMO. |
| 15:36.11 | ManxPower | rantsh: Perhaps you need to learn Asterisk before proceeding. Have you read The Book |
| 15:36.27 | anonymouz666 | [TK]D-Fender why do you think that? |
| 15:36.31 | rantsh | the one on the site, I've read quite a bit now |
| 15:36.53 | ManxPower | rantsh: if you cannot do something as simple as setting up 2 entries in sip.conf then you need to learn to do that before worrying about codecs. |
| 15:36.57 | cheshair | Mercestes, sure, i see... just a last question: i'm new to asterisk. i thought asterisknow could be a nice way to get into it. should i better install asterisk on my distro? |
| 15:36.59 | [TK]D-Fender | anonymouz666, because it overrides, reinvents the wheel, and then devalidates previous learning's application. |
| 15:37.08 | [TK]D-Fender | cheshair, YES |
| 15:37.08 | seele_ | hello I need a command to context change ... like DND *78 but for change the actual context to other |
| 15:37.09 | rantsh | but I don't have a REAL testing enviroment, so I must be sure of what I'm doing every time |
| 15:37.15 | seele_ | any suggest ? |
| 15:37.18 | ManxPower | rantsh: you CANNOT. |
| 15:37.26 | tzafrir | cheshair, if you're comfortable with a specific distro: certainly |
| 15:37.28 | [TK]D-Fender | cheshair, Ditch that ISO and install any decent plain distro and install * from source on it. |
| 15:37.29 | ManxPower | If you do not have a real testing enviroment then you cannot learn Asterisk |
| 15:37.33 | cheshair | [TK]D-Fender: i see, many thanks! |
| 15:37.58 | rantsh | what can I say.... don't tell me tell my boss :s |
| 15:38.09 | danp | [TK]D-Fender: cool, thanks. i'm looking to upgrade from 2.0.1 to 2.1.1 |
| 15:38.15 | [TK]D-Fender | cheshair, np, and when you are in a sane environment that you can control we'll be here to help you when you hit some bumps on the road |
| 15:38.16 | ManxPower | rantsh: many people use Asterisk for a YEAR before trying to deploy it into a production enviroment. |
| 15:38.31 | ManxPower | rantsh: I'm sorry, but I cannot help someone that has designed their project for failure. |
| 15:38.46 | rantsh | sorry dude, thanks anyway man |
| 15:39.16 | negativeduck | rantsh, you' |
| 15:39.22 | negativeduck | ve got to be able to build a test box. |
| 15:39.25 | rantsh | Iwon't give up though... |
| 15:39.31 | cheshair | [TK]D-Fender & tzafrir: well boys, you've got to understand me... i happened to see this "all-in-one" distro and i thought it was good... see you in an hour, many thanks! :-) |
| 15:39.35 | ManxPower | seele_: you cannot do that unless you want to transfer the call to a specific extension that uses a Goto to send the call to another context |
| 15:39.35 | negativeduck | grab a compuer off the shelf blow the dust out and play :) |
| 15:39.58 | [TK]D-Fender | cheshair, Sorry to disappoint.... they more it tries to do for you the less you can do yourself. |
| 15:40.09 | Mercestes | [TK]D-Fender, Asterisknow is a flaming pile of shit as far as I am concerned. =/ |
| 15:40.12 | [TK]D-Fender | cheshair, not a way to "learn" anything unfortunately. |
| 15:40.22 | ManxPower | seele_: Contexts are part of Asterisk's security design (such as it is) and it is not east to move between contexts during a call |
| 15:40.46 | [TK]D-Fender | ok, off to lunhc, back in many hours. |
| 15:40.55 | seele_ | ManxPower, ok, then how can I make a phone block system? ... to block outgoing call when I'm not in the office |
| 15:40.56 | cheshair | i see, see you later with my new asterisk |
| 15:41.01 | *** part/#asterisk cheshair (n=fabio@host-78-12-100-200.cust-adsl.tiscali.it) |
| 15:41.38 | ManxPower | seele_: Easy if you know asterisk. Not easy if you are a n00b |
| 15:41.59 | seele_ | ManxPower, I'm a n00b |
| 15:42.04 | ManxPower | seele_: perhaps you want to do this "prevent others from using my phone when I am not at my desk" |
| 15:42.17 | seele_ | ManxPower, yes |
| 15:42.17 | Jingles | seele_: use of 'GotoIfTime' functionality. |
| 15:42.20 | ManxPower | seele_: then forget about locking your phone and learn Asterisk first. |
| 15:42.47 | *** join/#asterisk ghento (n=ghento@CPE001124d2c50e-CM0011e6c416f1.cpe.net.cable.rogers.com) |
| 15:42.57 | seele_ | ManxPower, I'm learning asterisk ... I'm only asking for some help |
| 15:43.22 | seele_ | Jingles, no times ... I need a function like *78 |
| 15:43.34 | ManxPower | seele_: There are a couple of ways you can do this. I suggest just requiring all uses to enter an auth or billing code anytime they make a toll call. |
| 15:43.34 | Jingles | oh. |
| 15:43.49 | Mercestes | seele_, I was about to suggest Authenticate() |
| 15:44.06 | Jingles | seele_: so, you're trying to put your phone into a kind of 'locked' mode when you put in a * code. |
| 15:44.12 | Mercestes | seele_, You could also do some shell scripting and an extension to enable/disable your phone. |
| 15:44.27 | ManxPower | seele_: You can also dial an extension, enter a code and set/unset a value in ASTDB, then check for that value on all outgoing calls. |
| 15:44.27 | seele_ | ManxPower, is good Idea ... |
| 15:44.32 | seele_ | Jingles, yes |
| 15:44.48 | seele_ | Mercestes, That is the idea |
| 15:45.13 | ManxPower | seele_: every single suggestion assumes you know how to use extensions.conf and can set up the dialplan to handle this. |
| 15:45.20 | Mercestes | There you go, three ideas. |
| 15:45.52 | Jingles | Manx: now *that* it pretty slick thinking - the ASTDB suggestion. |
| 15:46.05 | Jingles | I'm betting I can find a way to put that to good use. ;) |
| 15:46.11 | seele_ | ok thanks I'm go to work with that |
| 15:46.30 | Mercestes | Jingles, Don't feed his ego too much. He might pop. ;) |
| 15:46.41 | Jingles | lols. *makes a note of it* |
| 15:46.47 | ManxPower | *78 is DO NOT DISTURB. It has nothing to do with locking your phone |
| 15:47.13 | Mercestes | ManxPower, right, that is what he is trying to do. |
| 15:47.18 | Mercestes | "Do not disturb my phone while I am away." |
| 15:47.19 | seele_ | ManxPower, only a example |
| 15:47.23 | Mercestes | >.> |
| 15:47.25 | ManxPower | Jingles: Um, the standard way to set ANYTHING dynamically globally is to use AstDB |
| 15:47.40 | ManxPower | Mercestes: No, he is not trying to DND |
| 15:47.50 | Mercestes | lol. I know. |
| 15:48.02 | robl^ | "unplug the phone" |
| 15:48.04 | Mercestes | Comeon...it was kinda funny. |
| 15:48.05 | Jingles | well, just because that's the 'standard' doesn't mean the nobs that set up the production environment before I got here *did that*. ;) so, I've never seen it in our dialplan. |
| 15:48.06 | ManxPower | He is trying to feed his paranoia about people using his phone when he is not there to guard his phone. |
| 15:48.16 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:48.20 | ManxPower | Mercestes: You don't have a sense of humor. You didn't now that? |
| 15:48.30 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:48.38 | seele_ | not my paranoia ... the client paranoia |
| 15:48.40 | mocker | instead of an auth code, have it record all calls. |
| 15:48.40 | seele_ | LOL |
| 15:48.42 | Mercestes | ManxPower, I think I have a hypersensitivity to humor, if I'm finding things funny you don't. |
| 15:48.49 | mocker | Then you can bust the people who are breaking the rules! |
| 15:49.03 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 15:49.27 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 15:49.32 | Mercestes | ManxPower, You did mean me and not you, right? I'm confused. =/ |
| 15:49.37 | Mercestes | again |
| 15:50.12 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 15:50.21 | Mercestes | hey, that reminds me. anyone know how I can disable the CFwdAll softkey on the Cisco 7960s? Can you do that? |
| 15:50.41 | ManxPower | Mercestes: specify your protocol |
| 15:50.47 | Mercestes | SIP |
| 15:50.56 | Mercestes | :) |
| 15:50.58 | rhombus | I have a few Zap channels that are mysteriously off-hook. Is there a way to put them on-hook without having to restart Asterisk? |
| 15:51.16 | ManxPower | rhombus: "softhangup" |
| 15:51.43 | rhombus | In 1.2? |
| 15:51.54 | tzafrir | rhombus, show channels |
| 15:52.15 | rhombus | There are no calls active. What I have is Zap channels that are off-hook. |
| 15:52.18 | tzafrir | are there channels to hang up? (sodft hangup, as ManxPower mentioned) |
| 15:52.20 | _VoiceMeUp_COM | maybe a soft hangup sip/blah* could be nice.. for those who dont send the hangups and call lots;) |
| 15:52.21 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 15:52.32 | _VoiceMeUp_COM | coz rtptimeout can cause probs |
| 15:53.00 | irule | I press 1 to listen to the first voicemail, then I get hung up, this is 1.4, is this a horrible bug? should I install the latest 1.2 to make this go away? or is there an alternative? |
| 15:53.07 | rhombus | When I do a zap show channel X, I get "Hookstate (FXS only): Offhook" |
| 15:53.12 | _VoiceMeUp_COM | or is there something like the ser/cisco sister/brother a reinvite message every XX sec.. like 600 sec.. to make sure convo is still good |
| 15:53.22 | rhombus | so nobody can call out on that channel |
| 15:53.47 | ManxPower | rhombus: Is the channel an FXS channel? |
| 15:54.00 | rhombus | Signalling Type: FXS Kewlstart |
| 15:54.07 | _VoiceMeUp_COM | cluecon special flightt .. http://www.youtube.com/watch?v=mZbZWgXruIY |
| 15:54.12 | rhombus | is that an FXS or FXO channel? |
| 15:54.26 | tzafrir | it's an FXO channel if it has FXS signalling |
| 15:54.55 | ManxPower | rhombus: does softhangup NOT work? |
| 15:55.02 | *** join/#asterisk anthm (n=anthm@h46088d27.area4.spcsdns.net) |
| 15:55.02 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:55.23 | rhombus | "No such command 'softhangup'." |
| 15:55.32 | _VoiceMeUp_COM | space in between |
| 15:55.35 | _VoiceMeUp_COM | soft hangup |
| 15:55.41 | _VoiceMeUp_COM | or maybe core soft hangup in 1.4 |
| 15:55.45 | rhombus | okay |
| 15:56.04 | _VoiceMeUp_COM | then tab to the rest.. |
| 15:56.40 | seele_ | where can I change the incoming CID format? |
| 15:56.44 | rhombus | ManxPower: Can i specify a specific Zap channel with that? |
| 15:56.50 | _VoiceMeUp_COM | on the inbound ? |
| 15:57.03 | el_4_jinete | ~quit |
| 15:57.04 | jbot | No! You quit! |
| 15:57.04 | el_4_jinete | quit |
| 15:57.06 | el_4_jinete | exity |
| 15:57.06 | el_4_jinete | exit |
| 15:57.08 | el_4_jinete | quit |
| 15:57.13 | Qwell[] | ... |
| 15:57.20 | Qwell[] | nub |
| 15:57.29 | rhombus | that was funny. |
| 15:57.37 | robl^ | that was SCARY |
| 15:57.47 | rhombus | well, it's funny after you get over the fear. |
| 15:57.53 | rhombus | some people are just impatient |
| 15:58.45 | rhombus | how are channels specified with 'soft hangup'? |
| 15:59.46 | robl^ | rhombus: welcome to my world. One of my duties at work is to maintain a document management system for a law firm. I love when users complain "I told it to open the 210MB document, with full color photos, 2 secs ago. It didn't do anything so I closed the app. What's wrong?" |
| 16:00.25 | rhombus | This is why I don't do work for law firms (among other reasons) |
| 16:00.48 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 16:03.26 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:04.33 | rob0 | ~exit |
| 16:04.33 | jbot | well, exit is what I want |
| 16:04.37 | rob0 | haha |
| 16:05.30 | rob0 | ~this |
| 16:05.30 | jbot | somebody said this was ibot is hearing me talk |
| 16:05.53 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 16:06.08 | ManxPower | rhombus: softhangup zap/4-1 |
| 16:06.12 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 16:06.16 | ManxPower | the same output as "show channels" usually |
| 16:07.55 | ManxPower | "Our credit card machine says 'no line'"!!!! "What do you want me to do about it, that line does not go thru any equipment of mine." |
| 16:08.31 | rhombus | ManxPower: This is what I get: asterisk1*CLI> soft hangup zap/3-1 |
| 16:08.31 | rhombus | zap/3-1 is not a known channel |
| 16:08.34 | Mercestes | ManxPower, Your non-response suggests my request is practically impossible..:( |
| 16:08.55 | ManxPower | rhombus: does Zap/3-1 show up in "show channels" |
| 16:09.14 | ManxPower | I don't run analog lines thru Asterisk |
| 16:09.15 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 16:11.02 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 16:11.10 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-202.range86-136.btcentralplus.com) |
| 16:11.26 | ManxPower | rhombus: does Zap/3-1 show up in "show channels" I won't ask again |
| 16:11.32 | rhombus | no, it does not |
| 16:11.36 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:11.38 | rhombus | I don't think I am making myself understood here |
| 16:11.45 | rhombus | the physical channel is in an off-hook state |
| 16:11.45 | ManxPower | rhombus: then you can't use softhangup. |
| 16:11.47 | *** join/#asterisk mrdigital (n=err@207-172-229-100.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 16:12.04 | ManxPower | rhombus: Is there a phone or a phoneline on that device? |
| 16:12.14 | rhombus | yes |
| 16:12.16 | rhombus | this worked on Saturday |
| 16:12.34 | rob0 | Well sure, everything works on Saturdays. ;) |
| 16:12.39 | ManxPower | If you have a phone line plugged into a phone port you will see that issue. You will also blow the port if the first time the phone line rings. |
| 16:12.48 | ManxPower | rhombus: perhaps you just need to restart asterisk |
| 16:12.52 | ManxPower | well, stop and start it. |
| 16:12.52 | rhombus | okay |
| 16:13.33 | robl^ | or reboot... sometimes the analog zaptel hard goes wonky if there were any type of power fluctuations... |
| 16:14.03 | ManxPower | rhombus: I just tested it. For FXO ports, the onhook/offhook is not valid |
| 16:14.31 | rhombus | that doesn't make any sense |
| 16:15.35 | ManxPower | I didn't say it made sense. I said that is what happens |
| 16:15.40 | rhombus | the inaccessible channels are off-hook, the working channels are on-hook -- until they are taken off hook by asterisk |
| 16:16.16 | rhombus | I think I'm going to switch to Sangoma :( |
| 16:16.32 | ManxPower | rhombus: reboot the damn pbx |
| 16:16.47 | rhombus | that is what I was trying to avoid, but okay |
| 16:17.03 | ManxPower | rhombus: then see what the state of those channels are |
| 16:17.26 | ManxPower | onhook/offhook is NOT related to your problem; |
| 16:17.37 | ManxPower | you will see the same hookstate after a reboot, but I'll bet it starts working.l |
| 16:18.00 | ManxPower | I had a similar problem with ports just locking up and not working until I rebooted. This is why I don't use Analog cards anymore |
| 16:18.22 | rhombus | I have not had this problem with Sangoma hardware, only with Digium hardware |
| 16:18.40 | ManxPower | I cannot comment on that. |
| 16:19.24 | ManxPower | rhombus: I assume you are using the latest version of zaptel for whatever major release you are using |
| 16:19.59 | rhombus | yes... |
| 16:21.32 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 16:22.12 | DarKnesS_WolF | rhombus: i think i had problem like that |
| 16:22.22 | rhombus | okay |
| 16:22.30 | DarKnesS_WolF | rhombus: it was that there is too many zaptel modules running and no hardware for it :-) so make sure to run teh correct module only |
| 16:22.48 | DarKnesS_WolF | rhombus: so i do zaptel zttranscode wctdm |
| 16:22.50 | DarKnesS_WolF | cuz i hat tdm400p |
| 16:22.57 | DarKnesS_WolF | have * |
| 16:23.13 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 16:26.06 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:28.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:30.19 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 16:30.25 | falz | I'm having a hell of a time with 1.4. upgraded from 1.2 the other day and now I always get "488 Not acceptable here" when xferring calls (between cisco 7960's). 100% of the information I can find is regarding codecs, so I set disallow=all and allow=ulaw for testing, sip headers confirm this, but still the issue. |
| 16:30.54 | falz | the strange part is that the phones can call each other, it's only when one hits the 'transfer' button on the phone (which gives the held-user music on hold) they instantly get hung up on after hearing a split second of music |
| 16:31.51 | ManxPower | falz: WHERE are you putting the disallow and allow lines |
| 16:31.58 | falz | [general] in sip.conf |
| 16:32.07 | falz | no allow/disallow in each users phone config, so it hsould be picking up on that |
| 16:32.28 | ManxPower | so what codec does sip debug show is being used. Maybe the PHONES are not allowing ulaw |
| 16:33.13 | falz | well, I've never really read a sip debug. the header looks like an smtp header (generally) so that's readable. seems to be two different spots. one is in the body, with a=foo |
| 16:33.20 | falz | the other may look like: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) |
| 16:33.40 | puzzled | hi |
| 16:33.47 | falz | the one above, when only permitting ulaw, does indeed only list ulaw. |
| 16:34.03 | falz | this line specificly: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) |
| 16:34.18 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:35.32 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 16:35.59 | ManxPower | falz: the Combined is what you care about. |
| 16:36.11 | falz | ok, thanks. |
| 16:36.11 | ManxPower | There are TWO legs of the call. What does the OTHER leg show? |
| 16:36.32 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 16:37.05 | ManxPower | falz: but you are seeing that both ulaw AND alaw are being allowed. This is bad. |
| 16:37.33 | falz | that was one of my debug lines when messing with that. most recently I did allow both. I can set it back to ulaw only and the same thing happens |
| 16:37.36 | falz | let me try and verify |
| 16:37.39 | *** join/#asterisk obnauticus (n=admin@c-71-59-162-60.hsd1.wa.comcast.net) |
| 16:38.00 | *** join/#asterisk HoverHell (n=hell@212.193.68.13) |
| 16:38.06 | ManxPower | I would prefer ACTUAL CURRENT output, not some random output generated when you are setting random options |
| 16:38.07 | obnauticus | Is there a good Online MeetMeAdmin thing |
| 16:38.19 | obnauticus | like where I can manage confrences online |
| 16:38.52 | falz | ype, regenerating stuff that's clearer. sec. |
| 16:39.51 | HoverHell | Hi, All! |
| 16:39.51 | HoverHell | How do I make MixMonitor in context, where it immediately jumps to extension (context for sip clients)? |
| 16:40.07 | ManxPower | HoverHell: you do not. Mixmonitor just starts recording the call then continues in the dialplan at the next priority |
| 16:40.21 | falz | ManxPower: ok, now we have Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) |
| 16:40.38 | HoverHell | I tried writing MixMonitor to beginning of context, but (according to debug) it doesn't get executed. |
| 16:40.39 | *** join/#asterisk Greenbox (n=Brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 16:40.42 | ManxPower | falz: All calls have 2 legs. You are only showing 1 leg of the call |
| 16:40.48 | falz | however, the lines leading up to that still say , Found description format PCMA for ID 8, etc |
| 16:40.55 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) [NETSPLIT VICTIM] |
| 16:41.08 | ManxPower | HoverHell: mixmonitor can only be run as part of an active call leg. |
| 16:42.39 | ManxPower | A call comes in, matches an "exten =>" line. You do the various things including mixmonitor before the Dial line to start the 2nd leg of the call |
| 16:44.29 | falz | ManxPower: agreed, that is one leg. however, there are no further lines similar to the one above before I get the "488 Not Acceptable Here" message |
| 16:44.53 | ManxPower | falz: put the full sip debug of a failed call on pastebin.ca |
| 16:44.56 | falz | now, this is happening when one puts a user on hold (by pressing the xfer button). the original call works. |
| 16:44.58 | falz | sure. |
| 16:45.49 | *** join/#asterisk myiagy (i=myiagy@201.31.20.47) |
| 16:46.09 | falz | ManxPower: http://falz.net/asterisk/sipdebug1.txt |
| 16:46.43 | falz | this is from the point AFTER phone 302 calls 303. phone 303 then presses transfer (which means hold) |
| 16:47.05 | falz | and blegh, I should have fixed up the moh stuff before pasting that, I removed that to remove that from the possibilities (poorly formatted mp3, etc) |
| 16:47.48 | ManxPower | falz: do you have quotes on the callerid= lines in sip.conf? |
| 16:48.21 | falz | nope! hmm |
| 16:48.37 | falz | actually my callerid format is the old way. let me fix it to the non-deprecated way |
| 16:49.00 | ManxPower | define old way and define new way |
| 16:49.03 | Dr-Linux | what's digium support number? |
| 16:49.12 | Dr-Linux | linux-me- ?? something i guess |
| 16:49.12 | falz | calleridnum/calleridname vs 'callerid' |
| 16:49.17 | ManxPower | callerid=Robert Dobbs <666> is the correct way. It did not change |
| 16:49.23 | ManxPower | no, sip.conf |
| 16:49.36 | ManxPower | unless you are setting it in extensions.conf |
| 16:49.51 | ManxPower | but that would be silly |
| 16:50.31 | ManxPower | since you didn't use pastebin, I can't tell you what LINE NUMBER the problem starts, but if you look one of the SDPs have NO codecs allowed. I think that is coming from the phone. |
| 16:50.34 | falz | which item in the phone's [] are you requesting? I dont have the name specified there at all, just username=303 (or some number without spaces) |
| 16:50.53 | ManxPower | falz: why are you setting the callerid in extensions.conf? |
| 16:51.13 | *** part/#asterisk ManxPower (n=manxpowe@42.sub-75-200-20.myvzw.com) |
| 16:51.15 | falz | I have a setcallerid line in there. not sure why, it's just always been there |
| 16:51.20 | *** join/#asterisk ManxPower (n=manxpowe@42.sub-75-200-20.myvzw.com) |
| 16:51.23 | ManxPower | oops |
| 16:51.24 | falz | so it's stayed. probably doesnt do anything and is redundant redundant |
| 16:51.56 | ManxPower | perhaps you need to simplify your dialplan if you are doing callerid stuff in the dialplan |
| 16:52.39 | falz | just two instances, removed. |
| 16:52.47 | ManxPower | the full extent of your dialplan for exten => 303 should be exten => 303,1,Dial(SIP/303) (or whatever sip.conf entry is for that phone) |
| 16:52.53 | Dr-Linux | anybody knows digium support number? |
| 16:53.03 | ManxPower | Dr-Linux: it's on the fucking web site |
| 16:53.20 | Dr-Linux | fucking web site ohhh :P |
| 16:53.37 | falz | ManxPower: that didnt make a difference for this specific problem, but if it was redundant, I'm always happy to remove unnecessary garbage. |
| 16:53.37 | Dr-Linux | got it from there: |
| 16:53.38 | Dr-Linux | 877.LINUX-ME (toll free) |
| 16:53.57 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 16:54.01 | ManxPower | see how easy that was and you didn't even have to waste everyone's time. |
| 16:54.14 | ManxPower | you did anyway, but you didn't have to. |
| 16:54.26 | robl^ | Digium has a porn website now? |
| 16:54.43 | ManxPower | falz: I think the issue is with the configuration of the phone. (i.e. the .cnf or .xml files on your tftp server) |
| 16:54.56 | *** part/#asterisk HoverHell (n=hell@212.193.68.13) |
| 16:55.11 | *** join/#asterisk ramindia_ (n=ramindia@202.63.96.133) |
| 16:55.14 | falz | ManxPower: note that this problem only came up when moving from asterisk 1.4 from 1.2. nothing else changed, conf files the same. |
| 16:55.22 | Greenbox | robl^, o.0 |
| 16:55.23 | falz | only began mucking with codec stuff to find a fix |
| 16:55.56 | falz | the dialplan for exten 303 is indeed as you note above: exten => 303,1,Dial,Sip/303|15 (then some voicemail stuff after) |
| 16:56.03 | ManxPower | falz: perhaps it is a 1.4 problem. I don't use 1.4 in production as it is not stable enough for my requirements. |
| 16:56.05 | falz | however, it doesnt get that far. I dont even dial. the caller is put on HOLD and the call drops |
| 16:56.21 | *** part/#asterisk ramindia_ (n=ramindia@202.63.96.133) |
| 16:56.25 | falz | 302 calls 303. they talk as a test. 303 puts 302 on hold (or vice-versa) and call drops, with sip headers I pasted |
| 16:56.30 | ManxPower | falz: it should not matter but you are not using the format I'm using |
| 16:56.38 | falz | with parens? |
| 16:56.43 | ManxPower | yes. |
| 16:57.00 | ManxPower | There was nothing mentioned in UPGRADE.txt about comma .vs. parens? |
| 16:57.23 | DarKnesS_WolF | anyone has CID problems with SNOM ? i get the caller ID from sip / iax but not from PSTN .. and it works with GIPTEL 200 ... any idea ? asterisk 1.2 |
| 16:57.38 | robl^ | falz: the | is no longer valid in 1.4 should be exten +> 303,1,Dial(SIP/303,15) |
| 16:57.40 | falz | ManxPower: not on the one it put in my system. |
| 16:57.54 | Sweeper | configure: error: C++ preprocessor "/lib/cpp" fails sanity check <-- wtf does this mean, when ./configureing *? |
| 16:57.57 | ManxPower | falz: now you know where to start |
| 16:57.57 | robl^ | err. exten => |
| 16:57.58 | falz | ok. |
| 16:58.05 | DarKnesS_WolF | Sweeper: distro ? |
| 16:58.05 | falz | will fix all of that junk up and re-test |
| 16:58.08 | falz | only 20ish extensions |
| 16:58.08 | Greenbox | Sweeper, you need g++ |
| 16:58.33 | falz | strange, I did read over upgrading, but didnt see that. either I'm retarted or this debian package didnt name it what I thought they should name it |
| 16:58.39 | DarKnesS_WolF | Sweeper: mandriva gcc-g++ pr gcc-c++ and in debian / ubuntu u need to install g++ |
| 16:58.50 | DarKnesS_WolF | robl^: i don't htink | removed from 1.4 |
| 16:58.53 | Sweeper | oahhh |
| 16:59.08 | ManxPower | I really need to get breakfast and get to work |
| 16:59.09 | Sweeper | see, I thought maybe it would have said "g++ not found" |
| 16:59.12 | DarKnesS_WolF | i reacall i uwas using it few days ago with authentication |
| 16:59.23 | ManxPower | Sweeper: Well that would be the LOGICAL thing 8-) |
| 16:59.25 | DarKnesS_WolF | Sweeper: what is ur distro ? |
| 16:59.27 | Greenbox | Sweeper, well, cpp is part of g++ |
| 16:59.33 | Sweeper | DarKnesS_WolF: centos 5 |
| 16:59.43 | Sweeper | installing... |
| 17:00.09 | DarKnesS_WolF | Sweeper: aww i hate centos :P but ur missing g++ like Greenbox said |
| 17:00.11 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 17:00.28 | falz | robl^ / ManxPower so anything that was piped ( | ) and not parens over the function command should likely be replaced with a comman and encased in parens (generally speaking) ? |
| 17:00.53 | ManxPower | falz: I don't know. I never use | or , format |
| 17:01.00 | ManxPower | I always use the "function" format. |
| 17:01.04 | DarKnesS_WolF | ManxPower: what do u use ? |
| 17:01.08 | ManxPower | i.e. use () and , |
| 17:01.19 | falz | most of the data from this came from a ported over dialplan from pre asterisk 1.2 |
| 17:01.24 | ManxPower | Dial(SIP/ldkfhsd;lkfhg) is what I use |
| 17:01.27 | falz | I've rewritten most of it, but just a little at a time |
| 17:01.29 | Hmmhesays | fun |
| 17:01.43 | DarKnesS_WolF | ManxPower: i see :-) |
| 17:01.50 | Sweeper | ManxPower: btw, I solved my realtime problems |
| 17:02.11 | DarKnesS_WolF | flai'll start to port my * soon from 1.2 to 1.4 but i know it will be hell so simply i'll not port it i'll get the requremnets and start from scratch much easyer |
| 17:02.11 | ManxPower | I'll start to use 1.4 when people stop reporting show stopping bugs for it. |
| 17:02.15 | Sweeper | I'm just gonna use OpenSER to do a db query, and send the appropriate context to * in a sip header |
| 17:02.32 | Hmmhesays | hmmm, what is the purpose of that sweeper? |
| 17:02.57 | Sweeper | Hmmhesays: the problem with realtime is that you have to have logic in the flatfile that points at a context in the db |
| 17:03.11 | Sweeper | I can use the input from the sip header to define the db context |
| 17:03.25 | Sweeper | so I don't have to reload every time I create a new context in the db |
| 17:03.32 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:03.45 | Hmmhesays | even think of using func_odbc to query for context? |
| 17:04.20 | Sweeper | plausible |
| 17:04.31 | Sweeper | we'll see how much of a PITA SER is |
| 17:04.38 | Sweeper | since I will probably need it eventually |
| 17:05.06 | DarKnesS_WolF | Sweeper: why u'll need SER ? |
| 17:05.15 | Sweeper | DarKnesS_WolF: speeeeeeeeed |
| 17:05.25 | DarKnesS_WolF | Sweeper: speed ? for what ? |
| 17:05.51 | Sweeper | so I don't have to buy 10 asterisk servers >.> |
| 17:05.59 | Hmmhesays | not enough info |
| 17:06.08 | DarKnesS_WolF | :-D |
| 17:06.26 | DarKnesS_WolF | Sweeper: how many SIP account u have ? |
| 17:06.34 | Sweeper | DarKnesS_WolF: none, at the moment :) |
| 17:06.43 | DarKnesS_WolF | Sweeper: one * is more than enough :P |
| 17:06.44 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 17:06.52 | Sweeper | pffft |
| 17:06.59 | Sweeper | plan for the future, mang |
| 17:07.09 | Sweeper | SER is very cheap, hardware wise |
| 17:07.20 | Sweeper | we'll see how much time it costs |
| 17:07.31 | DarKnesS_WolF | best luck :-) |
| 17:07.39 | Sweeper | but I can use SER's lcr module to load balance the IVR asterisk servers |
| 17:07.44 | Hmmhesays | if you're creating a pretty simple SIP setup its not bad |
| 17:08.01 | Hmmhesays | you can do that, or you can use dispatcher for a far easier setup |
| 17:08.15 | DarKnesS_WolF | one question if power went down... is there awaya to force TDM400P to fwd the call when it comes to a FXS port while the power is down ? |
| 17:08.36 | Sweeper | power is down..... |
| 17:08.46 | Hmmhesays | I'm fairly well versed in openser |
| 17:08.48 | Sweeper | how is it gonna do anything? :v |
| 17:08.56 | Hmmhesays | ups |
| 17:09.02 | DarKnesS_WolF | Sweeper: remember hte old 656K modems :-D? |
| 17:09.04 | DarKnesS_WolF | 56 ? |
| 17:09.09 | Sweeper | then power isn't down.... |
| 17:09.11 | DarKnesS_WolF | u plug a line and a phone |
| 17:09.18 | Sweeper | DarKnesS_WolF: that's just a jumper |
| 17:09.19 | DarKnesS_WolF | call comes in |
| 17:09.19 | DarKnesS_WolF | call comphone rings |
| 17:09.27 | DarKnesS_WolF | even power is down |
| 17:09.39 | DarKnesS_WolF | can't TDM400P do teh same jumper in case pwoer down ? |
| 17:09.47 | Hmmhesays | DarKnesS_WolF: I've used quintum products in the past for such events |
| 17:09.54 | Sweeper | you could make one yourself |
| 17:09.57 | Sweeper | it's called... |
| 17:10.03 | Sweeper | a SPLITTER~ |
| 17:10.09 | DarKnesS_WolF | haha |
| 17:10.14 | DarKnesS_WolF | tought about that |
| 17:10.20 | DarKnesS_WolF | thought |
| 17:10.25 | Sweeper | that's exactly whats on the modem |
| 17:10.28 | Hmmhesays | quintum fxs/fxo unit will do automatic passthru when shut off |
| 17:10.30 | DarKnesS_WolF | but i think TDM should do so |
| 17:10.41 | DarKnesS_WolF | mmm |
| 17:10.49 | DarKnesS_WolF | Hmmhesays: actually it's really very easy idea |
| 17:10.56 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 17:10.59 | DarKnesS_WolF | anywya gtg pray will be back later |
| 17:11.06 | Hmmhesays | DarKnesS_WolF: yes |
| 17:12.41 | Sweeper | Hmmhesays: dispatcher looks nice, although I also want to do real LCR for termination |
| 17:12.54 | Hmmhesays | dispatcher is just a good load balancer |
| 17:13.07 | Sweeper | mkar |
| 17:13.28 | Sweeper | probably simpler to use lcr for lcr and dispatcher for load balancing :) |
| 17:13.42 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 17:14.42 | colde | Hi, i have a register => line in my sip.conf, however..when i do "sip show registry" from the commandline it doens't show up |
| 17:15.11 | Sweeper | try sip show peers |
| 17:15.28 | Hmmhesays | should be under sip show registry |
| 17:15.36 | Hmmhesays | right syntax? |
| 17:17.08 | falz | ManxPower: I've got the dial() stuff converted, and as a test, put in a callerid= field in sip.conf as you specified above. still problem. (it didn't sound like any of this would have been the issue at all, anyhow). previously you were mentioning that you thought that SDP didn't see any codecs (I don't know what SDP is, honestly). was it this that you were referring to? Got unsupported a:fmtp in |
| 17:17.59 | falz | </cut off> Got unsupported a:fmtp in SDP offer Adding codec 0x4 (ulaw) to SDP |
| 17:19.24 | falz | or can anyone detail what happens when a 7960 puts someone on hold (either directly or awaiting a transfer) that could cause it to hang up? it's sending it music on hold via asterisk, then the hang up |
| 17:20.17 | colde | Sweeper: it is under sip peers |
| 17:20.38 | Sweeper | colde: and what does it say? |
| 17:20.43 | Dr-Linux | 40 min to go, digium support is now answering my call |
| 17:21.00 | colde | Hmmhesays: i should think so: register => 46931435:password@musimi.dk/201 |
| 17:21.05 | colde | Sweeper: unmonitroed |
| 17:21.11 | colde | unmonitored* |
| 17:21.23 | Sweeper | hmm |
| 17:23.33 | colde | any ideas? |
| 17:24.29 | ManxPower | falz: I think the problem is caused by the version of firmware in your phone. |
| 17:25.06 | ManxPower | colde: put the register in [general] |
| 17:25.07 | falz | hmm I wonder if cisco has anything past sip 8.6 |
| 17:25.44 | ManxPower | colde: then do a reload chan_sip.so |
| 17:26.41 | colde | ManxPower: of course |
| 17:26.44 | colde | will do |
| 17:26.47 | Hmmhesays | that looks right |
| 17:26.55 | Hmmhesays | do you have it in the [general] section of sip.conf? |
| 17:27.03 | Hmmhesays | I don't know if that makes any difference or not |
| 17:29.00 | falz | hmm 8.6 is the latest from cisco. blegh |
| 17:29.53 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 17:29.56 | Qwell[] | anybody got an efnet server they like? just need something off the top of your head |
| 17:30.10 | falz | so strange. have been using asterisk since 1.0.0 and have never had a problem |
| 17:30.12 | *** join/#asterisk anthm_mobile (n=anthm@000-446-926.area4.spcsdns.net) |
| 17:30.16 | Qwell[] | I realize just how incredibly off-topic that is, sorry |
| 17:30.28 | falz | I usually use irc.easynews.com |
| 17:30.30 | falz | or qeast.net |
| 17:30.42 | Qwell[] | didn't qeast merge with servercentral? |
| 17:30.50 | falz | hmm not sure. perhaps I havent used that in awhile |
| 17:30.52 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 17:31.01 | falz | I've got some hardcoded in my irc proxy and I guess i haven't looked at it for a few years :) |
| 17:31.08 | Qwell[] | either something is very seriously wrong with my xchat, or like every efnet server is down |
| 17:31.25 | falz | try irc.easynews.com. they're always full of excessive bandwidth and decent sysadmins |
| 17:31.37 | Qwell[] | not wanting to resolve/connect :( |
| 17:31.43 | DarKnesS_WolF | Qwell[]: why efnet :D? |
| 17:31.45 | DarKnesS_WolF | |
| 17:32.12 | falz | irc.easynews.com has address 69.16.172.2 |
| 17:32.16 | falz | according to host |
| 17:32.18 | aptura | I saw a web site that shows a graphical status representation of efnet it would be interesting if freenode had the same thing. |
| 17:32.20 | Qwell[] | yeah |
| 17:32.37 | Qwell[] | aptura: well, freenode "owns" (read: manages) all of their servers, so...eh |
| 17:32.40 | falz | you have ptr and ident and all that jaz? |
| 17:32.45 | falz | *jazz |
| 17:32.52 | Qwell[] | falz: no, it just isn't connecting at all |
| 17:33.02 | falz | sounds like we found the problem. your end :) |
| 17:33.16 | Qwell[] | ...note that I'm still on freenode :p |
| 17:33.37 | Qwell[] | that's bizarre, it must be xchat |
| 17:34.08 | DarKnesS_WolF | Qwell[]: thx i just connected and got banned :-s |
| 17:34.10 | DarKnesS_WolF | D-line |
| 17:34.22 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 17:34.38 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 17:34.38 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 17:34.51 | DarKnesS_WolF | Qwell[]: irc.choopa.ca <-- works with me |
| 17:34.53 | Qwell[] | stupid xchat |
| 17:35.06 | *** join/#asterisk dalfry (n=dalfry@70.89.177.109) |
| 17:35.21 | DarKnesS_WolF | Qwell[]: move to irssi :-) it's the best |
| 17:38.26 | mvanbaak | macIrssi |
| 17:39.13 | mvanbaak | and btw |
| 17:39.19 | mvanbaak | xterm is a gui as well ;) |
| 17:39.31 | Qwell[] | you know, I would use xterm if it had tabs |
| 17:39.42 | Qwell[] | erm, maybe I'm thinking eterm. whatever |
| 17:39.49 | mvanbaak | eterm, gterm |
| 17:40.00 | mvanbaak | or use a tabbed WM like ion3 |
| 17:40.02 | Qwell[] | it did everything I wanted - except tabs |
| 17:40.31 | mvanbaak | you can use screen for that |
| 17:40.42 | DarKnesS_WolF | Qwell[]: ok i did ask a stupid question and i would liek to ask again :-D now i have TDM400P when the power is down can i do a failover so all incoming calls on FXO gose to FXS ? or i'm dreaming ? |
| 17:41.16 | mvanbaak | yeah |
| 17:41.22 | mvanbaak | screen + irssi and mutt |
| 17:41.27 | _VoiceMeUp_COM | any way to choose source ip when doing an out call ? .. lets say you have 2 trunks.. A and B .. with each ips.. sip/exten@A.. |
| 17:41.27 | _VoiceMeUp_COM | and the box has 2 ips... |
| 17:41.28 | mvanbaak | and vim |
| 17:41.42 | _VoiceMeUp_COM | so to use a source ip of value X when dialing |
| 17:42.04 | ManxPower | DarKnesS_WolF: NO! |
| 17:42.11 | mvanbaak | _VoicePulse: that's not asterisk's job |
| 17:42.11 | Trevor_b | DarKnesS_WolF: Are you saying if power to building is out, route calls differently, or if power to PBX is out bridge across the ports? |
| 17:42.19 | mvanbaak | your routing should take care of that |
| 17:42.46 | _VoiceMeUp_COM | me ? |
| 17:42.47 | _VoiceMeUp_COM | hmm |
| 17:42.57 | ManxPower | _VoiceMeUp_COM: The OS should define what IP to use |
| 17:43.03 | MindTheGap | Corydon76-work, it was dead easy to resolve the "s" dst thing... As you said, just pass the ${EXTEN} to the macro, then within the macro an exten => s,1,Goto(${ARG1},1) and a pattern match that would match ARG1. What I was failing to understand is that one can have a goto() send an extension to a pattern match insite a macro, i thought a goto had to go to an exact extension... thanks... |
| 17:43.04 | ManxPower | Based on it's routing tables |
| 17:43.11 | _VoiceMeUp_COM | well.. lets say i wanna send to trunk A.. depending on what number is dialed i wanna use an ip |
| 17:43.27 | mvanbaak | huh ? |
| 17:43.33 | _VoiceMeUp_COM | i mean asterisk will always use first ip in the ifconfig ? |
| 17:43.43 | ManxPower | MindTheGap: ${ARG1} is evaluated BEFORE the rest of the dialplan happens |
| 17:43.45 | mvanbaak | you mean call1 using trunkA should use another source ip then call2 using trunkA |
| 17:43.46 | _VoiceMeUp_COM | ifconfig has 2 ips.. all routable |
| 17:43.48 | Corydon76-work | MindTheGap: there you go :-) |
| 17:43.58 | _VoiceMeUp_COM | i wanna use ip1 if ii want or ip 2 |
| 17:43.59 | mvanbaak | _VoicePulse: you should use iproute2 for that |
| 17:44.07 | DarKnesS_WolF | Trevor_b: yes exactl :-) |
| 17:44.25 | Trevor_b | DarKnesS_WolF: No, i asked you two different questions.... |
| 17:44.28 | ManxPower | _VoiceMeUp_COM: If the OS could route the call using either of the two IPS, then I imagine the first one will be used by default. AGAIN this is not an asterisk thing this is an OS thing |
| 17:44.35 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:44.39 | _VoiceMeUp_COM | yeah |
| 17:44.40 | _VoiceMeUp_COM | i get it |
| 17:44.51 | _VoiceMeUp_COM | but we could of used an array from the bindaddr.. |
| 17:44.53 | DarKnesS_WolF | Trevor_b: no the power to the building is down :-D |
| 17:45.03 | _VoiceMeUp_COM | bindaddr[1] = 1.2.3.4 bindaddr[2]=2.3.4.5 |
| 17:45.05 | Qwell[] | DarKnesS_WolF: no, use a splitter or something |
| 17:45.11 | Trevor_b | _VoiceMeUp_COM: BSD makes that simple, linux does not. |
| 17:45.11 | _VoiceMeUp_COM | then on each peer use a bindaddr[1] |
| 17:45.13 | _VoiceMeUp_COM | etc |
| 17:45.23 | ManxPower | _VoiceMeUp_COM: you can't specify what IP to use. |
| 17:45.23 | _VoiceMeUp_COM | then we could have vlans style mappings |
| 17:45.26 | DarKnesS_WolF | Qwell[]: thx :-) |
| 17:45.27 | _VoiceMeUp_COM | i know.. |
| 17:45.32 | _VoiceMeUp_COM | ;) but would of been fun no ? |
| 17:45.38 | _VoiceMeUp_COM | s/fun/practical/ |
| 17:45.38 | mvanbaak | use bsd |
| 17:45.39 | HarryR | Trevor_b: how is it any difference? |
| 17:45.42 | Qwell[] | _VoiceMeUp_COM: would "have"... |
| 17:45.52 | mvanbaak | bbl, food is here |
| 17:45.57 | _VoiceMeUp_COM | not touching bsd with a pole |
| 17:45.58 | Trevor_b | DarKnesS_WolF: You would have to tie your UPS into the PBX and cause a context change, or at least thats my assessment of how I would think about it> |
| 17:46.05 | Trevor_b | harryr: What?\ |
| 17:46.21 | HarryR | uh, BSD making biding multiple ip addresses easier? |
| 17:46.36 | ManxPower | _VoiceMeUp_COM: bindaddr only works for the SIGNALLING, not the audio |
| 17:46.52 | *** join/#asterisk UKCoder (n=UKCoder@h-72-244-53-172.snfccasy.covad.net) |
| 17:46.59 | _VoiceMeUp_COM | ok |
| 17:47.00 | _VoiceMeUp_COM | then |
| 17:47.08 | Trevor_b | because BSD (open) uses PF and it has the ability to define a route with a gateway, then use certain traffic on that route, setting return addresses properly. |
| 17:47.19 | ManxPower | _VoiceMeUp_COM: you CANNOT do what you want to do in the way you want to do it. FIND ANOTHER WAY |
| 17:47.22 | _VoiceMeUp_COM | how about.. pishing a call to asterisk that has multiple addys.. will it answer from same ip it recieved on ? or choos ea random /routed one |
| 17:47.35 | *** part/#asterisk allen__s (n=chatzill@72.242.225.99) |
| 17:47.38 | HarryR | Trevor_b: and it wouldn't just be easier to make Asterisk bind to multiple ip addresses? :) |
| 17:47.45 | _VoiceMeUp_COM | asterisk does bind |
| 17:47.45 | Trevor_b | harryr: Works MUCH simpler then trying to do the same thing in Linux. |
| 17:47.47 | *** join/#asterisk pulu (i=pulu@wsip-68-98-213-162.ph.ph.cox.net) |
| 17:47.48 | Trevor_b | NO |
| 17:47.51 | _VoiceMeUp_COM | just problem is the reply |
| 17:47.52 | UKCoder | Hi all.... I know this is a long shot, but is there anyone here that attended the (SF) Bay Area AUG at the start of this year in Sunnyvale, CA, or anyone who knows the organizers there? :) |
| 17:47.56 | ManxPower | _VoiceMeUp_COM: If you are not using bindaddr then it should respond on the same IP it came in on |
| 17:48.00 | Trevor_b | harryr: How do you make asterisk decide to route out an IP that has no Gateway??? |
| 17:48.04 | ManxPower | But AGAIN this is an OS issue. |
| 17:48.15 | HarryR | uh, I mean just binding to the ip? |
| 17:48.30 | HarryR | you create & bind the socket, then just stick it in whatever event loop you already have |
| 17:48.33 | Trevor_b | Thats like ManxPower says, dont bind and it should return on the inbound IP. |
| 17:48.38 | _VoiceMeUp_COM | ManxPower doesnt.. |
| 17:48.45 | _VoiceMeUp_COM | i used bindaddr=0.0.0.0 |
| 17:49.00 | ManxPower | HarryR: that works if you only want to use 1 IP for Asterisk,. This lunatic wants to bind to multiple IPS. |
| 17:49.05 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 17:49.11 | HarryR | ManxPower: yah thats what I'm talking about |
| 17:49.12 | Trevor_b | damn client. |
| 17:49.19 | _VoiceMeUp_COM | sent the invite to .152 cameback from master ip |
| 17:49.19 | ManxPower | _VoiceMeUp_COM: bindaddr=0.0.0.o is the same as not having a bindaddr |
| 17:49.23 | HarryR | I've had a few times when I've needed 1 asterisk box binding to multiple ips |
| 17:49.31 | HarryR | for example, bridging lan to wan |
| 17:49.41 | falz | ManxPower: tested on some grandstream phones with my hang-when-on-hold, same issue there, so it's definitely asterisk |
| 17:49.45 | *** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 17:49.47 | HarryR | and other wierd configs like that |
| 17:49.49 | ManxPower | _VoiceMeUp_COM: I don't know how many times I can say "It is an OS thing". |
| 17:49.56 | _VoiceMeUp_COM | yep.. so invite 1.2.3.4 came back from 3.4.5.6 |
| 17:49.58 | _VoiceMeUp_COM | with a 100 trying |
| 17:49.59 | ManxPower | falz: downdrage to 1.2 |
| 17:50.06 | _VoiceMeUp_COM | ok thanks |
| 17:50.06 | Trevor_b | HarryR: My * has 2 IP's, one for WAN inbound and 1 for internal VoIP LAN, i never see it respond on the wrong ip... |
| 17:50.07 | UKCoder | Anyone in here know of a reputable asterisk consultancy in northern california? |
| 17:50.10 | falz | ManxPower: sounding like the only option. |
| 17:50.22 | _VoiceMeUp_COM | tough you said ; [13:47] ManxPower: _VoiceMeUp_COM: If you are not using bindaddr then it should respond on the same IP it came in on |
| 17:50.23 | _VoiceMeUp_COM | ;) |
| 17:50.25 | falz | but that never satisfies anything, but coincides withw hat you were saying- 1.4 is buggy |
| 17:50.25 | ManxPower | Trevor_b: that is because you allow the OS to figure it out. |
| 17:50.27 | Trevor_b | but those are seperated networks, doesnt need a gateway for internal |
| 17:50.32 | falz | I'm posting to asterisk-users anyhow |
| 17:50.50 | Trevor_b | ManxPower: right, the OS ALWAYS decides now to route traffic, not the application. |
| 17:50.51 | ManxPower | _VoiceMeUp_COM: I was assuming that your IPs were in DIFFERENT SUBNETS. Apparently they are not. |
| 17:50.52 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 17:51.03 | _VoiceMeUp_COM | Trevor_b also the intenral is filtered by the external ip and lcoalnet settings |
| 17:51.06 | _VoiceMeUp_COM | this is wy |
| 17:51.12 | _VoiceMeUp_COM | yes |
| 17:51.14 | _VoiceMeUp_COM | diff subs |
| 17:51.18 | _VoiceMeUp_COM | got 5 ips on box |
| 17:51.18 | ManxPower | _VoiceMeUp_COM: Are you trying to design asterisk to NOT work |
| 17:51.27 | _VoiceMeUp_COM | does 2 are 209.xxxx and 72.xxx |
| 17:51.35 | ManxPower | _VoiceMeUp_COM: and each of those IPs are on different subnets? |
| 17:51.47 | _VoiceMeUp_COM | FLASE [13:50] ManxPower: _VoiceMeUp_COM: I was assuming that your IPs were in DIFFERENT SUBNETS. Apparently they are not. |
| 17:51.54 | _VoiceMeUp_COM | cant be fuirther apart |
| 17:52.09 | ManxPower | _VoiceMeUp_COM: then your network are really fucked up. |
| 17:52.17 | ManxPower | Or, perhaps you are running NAT |
| 17:52.19 | Trevor_b | _VoiceMeUp_COM: are you trying to get traffic to 209.x.x.x to return on that circuit and those to 72.x.x.x to return out on that seperate internet circuit?? |
| 17:52.22 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 17:52.38 | *** join/#asterisk angryuser (n=Miranda@df01t2-212-195-198-141.d4.club-internet.fr) |
| 17:52.44 | ManxPower | _VoiceMeUp_COM: "route -an" |
| 17:52.47 | _VoiceMeUp_COM | well .. ser pushed to asterisk on 72..x.x.x.x. asterisk responded from 209.. |
| 17:52.49 | _VoiceMeUp_COM | k |
| 17:52.50 | ManxPower | that tells you the OS routing |
| 17:53.02 | ManxPower | _VoiceMeUp_COM: what is the netmask used? |
| 17:53.04 | _VoiceMeUp_COM | yes |
| 17:53.05 | _VoiceMeUp_COM | thanks |
| 17:53.15 | Trevor_b | _VoiceMeUp_COM: was SER in the same 72.x.x.x subnet? Sounds like it wasnt. |
| 17:53.26 | _VoiceMeUp_COM | 209.172.58.96 0.0.0.0 255.255.255.224 U 0 0 0 eth0 |
| 17:53.26 | _VoiceMeUp_COM | 72.55.158.128 0.0.0.0 255.255.255.128 U 0 0 0 eth0 |
| 17:53.28 | _VoiceMeUp_COM | ;) |
| 17:53.46 | _VoiceMeUp_COM | then a 0.0-.0.0 for rest on 209.... |
| 17:53.50 | Trevor_b | _VoiceMeUp_COM: did you setup the tables from the advanced linux routing howto? |
| 17:53.56 | _VoiceMeUp_COM | no |
| 17:54.00 | Trevor_b | thats your problem |
| 17:54.01 | _VoiceMeUp_COM | didnt touche tables |
| 17:54.12 | _VoiceMeUp_COM | just added an alias |
| 17:54.16 | Trevor_b | you CANNOT define 2 default gateways and assume the OS will just do its thing..... |
| 17:54.16 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 17:54.24 | _VoiceMeUp_COM | theres 1 gw |
| 17:54.26 | _VoiceMeUp_COM | not 2 |
| 17:54.30 | _VoiceMeUp_COM | its same router that does it |
| 17:54.30 | ManxPower | he does not have ANY default gateways defined |
| 17:54.36 | *** join/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 17:54.39 | ManxPower | _VoiceMeUp_COM: NO! |
| 17:54.39 | _VoiceMeUp_COM | 0.0.0.0 209.172.58.97 0.0.0.0 UG 0 0 0 eth0 |
| 17:54.47 | _VoiceMeUp_COM | just pasted first 2 here the last |
| 17:54.51 | _VoiceMeUp_COM | default GW |
| 17:54.57 | ManxPower | _VoiceMeUp_COM: and what is the IP address of the SER box? |
| 17:54.57 | Trevor_b | sorry read that backwards |
| 17:55.01 | _VoiceMeUp_COM | that router.. .97 deals with both routes |
| 17:55.07 | _VoiceMeUp_COM | not in there |
| 17:55.11 | _VoiceMeUp_COM | in 209 |
| 17:55.13 | _VoiceMeUp_COM | oh |
| 17:55.16 | _VoiceMeUp_COM | ah |
| 17:55.38 | _VoiceMeUp_COM | so the routing says.. well it came from a 209 for a 72.. . we have routing for a 209 only send it back via that ? |
| 17:55.44 | ManxPower | No., |
| 17:55.47 | Trevor_b | _VoiceMeUp_COM: read the linux advanced routing howto, you need to setup custom tables and gateways to allow 2 diverse paths out. |
| 17:55.49 | _VoiceMeUp_COM | argh.. |
| 17:55.50 | _VoiceMeUp_COM | ok |
| 17:55.54 | ManxPower | What that is saying is "what is the IP address of the SER box" |
| 17:56.09 | _VoiceMeUp_COM | ok well its in the 209.x |
| 17:56.14 | ManxPower | and that is what I am also asking. |
| 17:56.15 | _VoiceMeUp_COM | for now |
| 17:56.19 | _VoiceMeUp_COM | ill add another in 82 |
| 17:56.21 | _VoiceMeUp_COM | 72 |
| 17:57.03 | ManxPower | _VoiceMeUp_COM: it is pretty obvious you don't know much about networking or you would understand just how important it is to know the netmasks and the IPs of both devices. |
| 17:57.32 | _VoiceMeUp_COM | reading that how to ..thanks |
| 17:58.12 | ManxPower | For example of your asterisk box was on 209.172.58.70 and your SER was 209.172.58.130 then there is your problem right there, assuming your netmask is 255.255.255.224 |
| 17:58.44 | _VoiceMeUp_COM | http://lartc.org/howto/ |
| 17:58.45 | _VoiceMeUp_COM | reding |
| 17:58.55 | ManxPower | There wouldbe no issue if your netmask was 255.255.255.0 of course. |
| 17:59.10 | lesouvage | Is there a command to find out what digium card is used in the server |
| 17:59.21 | _VoiceMeUp_COM | hmm |
| 17:59.26 | Trevor_b | ManxPower: let him read that howto, after that questions should be better phrased ;) |
| 17:59.27 | ManxPower | Trevor_b: I don't know if he is trying to have two gateways |
| 17:59.41 | _VoiceMeUp_COM | ser .11x asterisk .11x nextmask 224 |
| 17:59.48 | ManxPower | lesouvage: dmesg |
| 17:59.55 | _VoiceMeUp_COM | netmask is ok mate |
| 18:00.00 | _VoiceMeUp_COM | for the 209 |
| 18:00.00 | b1shop | how's the Polycom Soundstation IP 4000 for a conference phone? |
| 18:00.30 | Qwell[] | b1shop: polycom conf phones are pretty great |
| 18:00.40 | ManxPower | _VoiceMeUp_COM: That SHOULD be OK. |
| 18:00.41 | UKCoder | b1shop: great, I use one here at the office |
| 18:01.01 | b1shop | sweet. wanted some feedback before i clicked submit |
| 18:01.03 | b1shop | ;-) |
| 18:01.16 | b1shop | pricey though... |
| 18:01.37 | Qwell[] | for good reason |
| 18:01.42 | Trevor_b | ManxPower: ahh, Yeah if its all local on the second subnet then its exactly how im using mine, but yes netmask is VERY important. |
| 18:01.50 | colde | when doing a dial to a SIP connection, how do i dial a specific number on that connection? |
| 18:01.53 | ManxPower | Trevor_b: I'll bet the only thing he'll get out the howto is just how much he does not know. |
| 18:02.02 | Trevor_b | hehe |
| 18:02.24 | falz | ManxPower: works after 1.2 downgrade. will stick with this for awhile. |
| 18:02.45 | ManxPower | falz: I don't care what Digium says. 1.4 is not ready for production |
| 18:03.06 | ManxPower | colde: Dial(SIP/destinationumber@sipconfentry) |
| 18:03.08 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:03.22 | ManxPower | Networking is HARD. |
| 18:03.35 | colde | ManxPower: cool |
| 18:03.41 | ManxPower | The WAN that I manage has something like 300 subnets on it. |
| 18:03.48 | falz | I used to be more skeptical with that stuff, since apache 2.2, mysql 5.0, php 5.2 all work decently. now to go back to that mode |
| 18:04.04 | lesouvage | ManxPower: dmesg gives me a couple of hunderds of this lines: IN-inside:IN=eth0 OUT= MAC=00:18:8b:85:4f:61:00:11:5c:f2:52:af:08:00 SRC=192.168.2.88 DST=192.168.0.240 LEN=369 TOS=0x00 PREC=0x00 TTL=125 ID=28103 PROTO=UDP SPT=1000 DPT=5060 LEN=349 |
| 18:04.20 | falz | those look like just firewall logs |
| 18:04.25 | _VoiceMeUp_COM | they are ;) |
| 18:04.34 | ManxPower | lesouvage: the information is put in /var/log/messages when the driver is loaded. |
| 18:04.52 | lesouvage | thanks |
| 18:05.03 | *** part/#asterisk UKCoder (n=UKCoder@h-72-244-53-172.snfccasy.covad.net) |
| 18:07.03 | ManxPower | lesouvage: The information may not provide information about the hardware revision |
| 18:10.33 | *** join/#asterisk Waverly360 (n=Waverly3@209.12.249.243) |
| 18:11.44 | Waverly360 | Has anyone here ever tried connecting an asterisk box to a Cisco Router via a T1 cable through a PRI card in the asterisk box, and a T1 card in the router? |
| 18:13.28 | falz | you'd need to demux it, it wouldnt work properly. |
| 18:13.33 | falz | er wait, you mean a direct xover connection? |
| 18:13.34 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 18:13.41 | Waverly360 | Yeah |
| 18:13.46 | falz | make a crossover t1 cable and it will work |
| 18:13.49 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 18:14.02 | falz | do digium pri cards have built in csu/dsu? |
| 18:14.09 | Waverly360 | Well, I got the connection to synch properly |
| 18:14.24 | Waverly360 | I don't know much about it..I was using a sangoma card |
| 18:14.58 | falz | what are you trying to accomplish, exactly? |
| 18:15.30 | Waverly360 | Here's my dilemma exactly. There are several offices with old Toshiba PBXes. |
| 18:16.13 | Waverly360 | Each toshiba has a t1 coming off of it that connects into a cisco router, which routes the voice via vpn to another PBX based on the 5 digit number that's passed across it. |
| 18:17.15 | Waverly360 | So the connection is PBX <-> t1 <-> cisco 1700 <-> vpn across the internet <-> cisco 2600 which is the central hub <-> local pbx |
| 18:17.32 | Waverly360 | sorry, there's another t1 connection in between the last two. |
| 18:17.50 | Waverly360 | I want to replace one of the toshiba's with an asterisk box, and make it work the same way. |
| 18:18.15 | falz | it sounds like you're overthinking things. if you have multiple t1's just send the voice traffic over as IP data, instead of some funky direct connection |
| 18:18.16 | Waverly360 | The t1 is setup to do E&M Immediate signalling, b8zs, esf |
| 18:18.51 | Waverly360 | I can't change the way it's setup now |
| 18:18.53 | falz | especially if you have two, you can either do mlppp to bond them w/ redundancy, or dedicate one to certain traffic if you want |
| 18:19.00 | falz | but the toshiba's are leaving? |
| 18:19.09 | Waverly360 | well, they want to add a new one |
| 18:19.20 | Waverly360 | but I'm trying to talk them into an asterisk box instead of a toshiba |
| 18:19.26 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.65) |
| 18:19.28 | zeeesh | hi |
| 18:20.00 | Waverly360 | but they want it setup the exact same way..using the cisco's to route the calls based on a 5 digit number..the first two digits stand for the office code, and the last three are the extension to be dialed |
| 18:20.12 | Waverly360 | I put one in place to test it out, and got the connections to synch up |
| 18:20.52 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 18:22.15 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:26.26 | *** join/#asterisk anthm (n=anthm@000-457-779.area4.spcsdns.net) |
| 18:26.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:30.41 | Innatech | who wants to reassure me that the switches in the polycom phones are reliable and I don't need redundant cabling? |
| 18:31.19 | Waverly360 | but when I try to do a dial command into the cisco connection it just sits there. I get no dialing tones..just silence...I don't know..I was just hoping someone else had tried something similar, so I could have a starting place. |
| 18:33.05 | *** part/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 18:33.21 | *** join/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 18:34.05 | Innatech | that's kind of a strange configuration. You're trying to retrofit Asterisk onto someone's legacy kludge. Time to bite the bullet and reconfig the T1s, I'd think. |
| 18:34.13 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 18:34.22 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 18:34.39 | Innatech | Tell whoever's counting the beans that it'll mean less cost and pain downt the road./ |
| 18:34.51 | falz | ManxPower: looks like someone posted the fix for me. simple one liner. http://lists.digium.com/pipermail/asterisk-users/2007-June/190741.html |
| 18:35.01 | skymeyer | evening all, if i want to compare both asterisk and cisco call manager, what are the major pros and conns of both ? |
| 18:36.01 | Innatech | well, there's the standard choice between open and proprietary models, for starters.... |
| 18:36.41 | skymeyer | Innatech: it is, but i'm more interested in feature advantages / disadvantages, i dont have many call manager experience, only asterisk and trixbox |
| 18:37.46 | Innatech | Well, then you likely know as much as I do. My experience is with * and trix as welll. My take on it is that if you're comfortable with Asterisk than you don't need to worry about CallManager--you don't need the Cisco security blanket. |
| 18:38.01 | Innatech | and the associated taxes... |
| 18:38.03 | obnauticus | Is there any online GUI for MeetMeAdmin confrence administrator. |
| 18:38.34 | skymeyer | Innatech: indeed, i follow your points made ;) but our customers are blinded by the cisco logo though |
| 18:38.45 | skymeyer | Innatech: thnx anyway ;) |
| 18:39.12 | *** join/#asterisk gardo (n=gardo@121.97.196.53) |
| 18:41.06 | obnauticus | Is there any way I can umm |
| 18:41.14 | obnauticus | Manage my asterisk server through a different web server. |
| 18:41.19 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:41.23 | obnauticus | Like with FreePBX and ARI, but on a different web server. |
| 18:41.26 | Innatech | Sure! I actually find that you can tell the client exactly that--and let them decide if they want to pay more for a packaged product from a major vendor. Some of them really do want that, even when it's made plain that it's the only difference. |
| 18:41.45 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 18:44.03 | *** join/#asterisk saftsack (n=saftsack@pD9E05EE2.dip.t-dialin.net) |
| 18:46.06 | Innatech | Can anyone reassure me that using VLANs with the switches built into Polycom phones instead of installing separate voice and data ethernet drops per desk will provide acceptable performance? |
| 18:46.50 | falz | probably depends on how much traffic you're using. video editing or anything like that? |
| 18:47.01 | Innatech | nah, standard office applications plus voice. |
| 18:47.17 | *** join/#asterisk nicchap (n=nicchap@216.209.85.2) |
| 18:47.40 | nicchap | Good day * users. Has anyone experienced the following message out there? ERROR[770] app_dial.c: Could not stop autoservice on calling channel |
| 18:48.03 | falz | I don't have polycom's, but I use the voice vlan feature of cisco phones with a built in switch, works fine for normal stuff, have never had an issue |
| 18:48.12 | Innatech | I've just not had a lot of experience with the Polycoms, and I'm not sure if I'm safe assuming everything will work transparently. We're cabling now, so if I want redundancy this is the time to put it in. |
| 18:48.22 | skymeyer | Innatech: got some bad performance on some voip phones using VLAN, others are working great, but no polycom experience here |
| 18:50.07 | Innatech | hmmm. Interesting..... |
| 18:51.17 | Innatech | there's no PoE on the polycom's switched port, right? That'd be one concern. Heh. |
| 18:52.58 | Innatech | At least in a room w/more than one phone. |
| 18:54.19 | skymeyer | Innatech: i shouldnt worry about performance when using "proper" phones (i suspect polycom is :-), i had some terrible experience only with low-end (= cheap) phones |
| 18:55.26 | Innatech | yeah, that's my sense too. Still, I always get that nervous feeling when the cabling guy asks me to make sure I have what I need. |
| 18:56.03 | skymeyer | they just want to make money ;) |
| 18:56.05 | skymeyer | lol |
| 18:56.46 | aptura | What is the normal range of POE? Can it go beyond the 100 meters? |
| 18:56.58 | Innatech | yeah, there's that too. Heh. Still, the old cabling at this place was scary. CatIII. Ugly splices. I don't want to wind up falling back onto it for data. |
| 18:57.37 | skymeyer | use phones with gigabit switchports installed to be sure ;) |
| 18:58.35 | tzafrir_home | obnauticus, where exactly does FreePBX assume that Asterisk runs on the same server? In very few places it writes directly to the config files |
| 18:58.42 | Innatech | heh. Not likely. I had to twist arms to keep the client from going entirely with Snom300s. |
| 18:59.14 | aptura | Innatech what line of work is the client involved in. |
| 18:59.36 | Innatech | law. |
| 18:59.50 | Innatech | they're spend a lot of time on the phone. |
| 18:59.57 | Innatech | *spending |
| 19:00.05 | *** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 19:00.08 | *** mode/#asterisk [+o Corydon76-work] by ChanServ |
| 19:00.18 | aptura | Attorneys can be demanding. |
| 19:00.20 | aptura | :) |
| 19:00.33 | aptura | Hope you picked a good one. |
| 19:00.35 | obnauticus | tzafrir so what does that mean? |
| 19:00.43 | obnauticus | like.. |
| 19:00.50 | obnauticus | how would i like the config files over the internet... |
| 19:00.51 | obnauticus | or lan |
| 19:00.52 | Innatech | Yeah, they're good people. And you're right, attys can be a pain. Most of my clientele is law firms. |
| 19:01.31 | Innatech | They do understand the princple that you should usually take the advice you're paying for, though, which I appreciate. |
| 19:02.22 | Innatech | Doctors, on the other hand..... {runs away} |
| 19:03.00 | skymeyer | Innatech: you all using snom phones ? |
| 19:03.35 | Innatech | Well, for this client I've convinced them to use them only at guest extensions and for auxillary desks/workers. |
| 19:03.58 | Innatech | The clerical staff will get poly 430s and the attorneys will get 501s. |
| 19:03.59 | skymeyer | aptura: pay attentention with PoE with cables longer than 100 meters |
| 19:04.33 | skymeyer | Innatech: the snoms we have deployed are working great, but we are mostly satisfied with the cisco phones |
| 19:05.53 | Innatech | skymeyer: I'm happy with the snoms I'm using in house, and all of the leaner clients I've set up on snom have been happy. But, those clients had less to work with and haven't been spoiled by Nortel. |
| 19:06.04 | skymeyer | :) |
| 19:07.50 | *** join/#asterisk joe-f (n=joef@c-71-201-188-239.hsd1.il.comcast.net) |
| 19:08.42 | *** join/#asterisk Wag3Slav3 (n=gregg@71-32-119-21.bsmr.qwest.net) |
| 19:09.38 | Wag3Slav3 | I have a couple of 800 numbers that never ring, they just auto answer to a voice prompt. My Asterisk doesn't detect this as a connection, it waits for 30 seconds then opens the native link. Can someone help me on what is causing this? |
| 19:10.15 | Wag3Slav3 | All other numbers work fine, and this is a SIP channel link |
| 19:10.25 | _VoiceMeUp_COM | http://lartc.org/howto/lartc.rpdb.html |
| 19:10.40 | _VoiceMeUp_COM | is that thes section i need to apply to use source ip calling ? |
| 19:12.25 | Mercestes | Innatech, Polycoms are great phones, 601s inherently support PoE, the others require a special dongle cable. My advice, fi you wanna use PoE on a polycom, go with the 601. As far as the secondary port? It does not provide PoE. |
| 19:14.37 | Wag3Slav3 | Is there a specific term for that never ring instant connect thing that I can google for to get this fixed? I think it's a connection detection problem, but I can't seem to find anything about it. |
| 19:15.12 | b1shop | OT.. but anyone have recommendations for server room environmental monitoring? We blew our AC unit the other night and almost lost a ton of gear. |
| 19:15.42 | falz | a ghetto method is to monitor snmp values from routers and such |
| 19:15.56 | lilalinux | I'm trying to install 2 hfc cards, 1 in nt-mode, the other in te mode. I always get the error "chan_zap.c:1097 zt_open: Unable to specify channel 1: Invalid argument" and in kernel "ioctl32(asterisk:20267): Unknown cmd fd(17) cmd(40044a26){00} arg(ffe37988) on /dev/zap/channel" |
| 19:16.00 | falz | simple script to read it and email you if there's an issue, or use something with snmp to graph it |
| 19:18.43 | *** join/#asterisk BillieGDJoe (n=fer@189.4.62.71) |
| 19:18.48 | BillieGDJoe | hi folks |
| 19:19.28 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 19:19.29 | holiday_42 | b1shop: seen a place using "sensaphone" products for temperature... might be overkill though |
| 19:20.58 | BillieGDJoe | when I transfer a call to a line, and this is busy, the call is finished. how I make this go back to line who transfer the call ??? |
| 19:21.08 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 19:21.14 | Innatech | mercestes: could you elaborate on your preference for the 601s over the 501s? The 430 does not require a PoE dongle, AFAIK, however the 501 does. |
| 19:21.38 | *** join/#asterisk cheshair (n=fabio@host-78-12-100-200.cust-adsl.tiscali.it) |
| 19:22.24 | kombi | I'm having this weired problem with meetme. After solving all the zaptel and ztdummy issues, I get no more errors even with verbosity cranked all the way up. But when I call into a conference, CLI says "plaging 'conf-onlyperson'" while nothing can be heard. The call gets stuck in there even after hanging up and asterisk can only forcefully be stopped.. |
| 19:22.42 | b1shop | holiday_42: yeah. that may be overkill |
| 19:23.16 | Mercestes | Innatech, more buttons, better voice enhancement software, does not require a PoE dongle, which sounds like the 430's will cover that for you. |
| 19:23.28 | Mercestes | Innatech, that dongle is the #1 concern. |
| 19:24.17 | Mercestes | kombi, Is thsi a new install? |
| 19:24.40 | Innatech | mercestes: great, thanks. |
| 19:24.46 | Mercestes | Innatech, np. |
| 19:24.48 | Mercestes | And remember! |
| 19:24.50 | Mercestes | ~phones |
| 19:24.50 | jbot | i guess phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 19:24.55 | kombi | Mercestes: well, re-installed quite a few times to wrangle zaptel, but appart from that no |
| 19:25.05 | Mercestes | kombi: Does outbound work ok? |
| 19:25.14 | kombi | yip |
| 19:25.23 | kombi | so does inbound and moh |
| 19:25.25 | Mercestes | hrm, that *is* weird then |
| 19:25.46 | Mercestes | What version of * and what type of phones? |
| 19:25.58 | kombi | 1.4 and xlite |
| 19:26.04 | Mercestes | Nat? |
| 19:26.18 | kombi | 1.4.5 to be precise |
| 19:26.30 | Mercestes | I heard 1.4.5 was pretty shitty |
| 19:26.36 | Mercestes | only a slight improvement over 1.4.4 |
| 19:26.47 | cheshair | hi guys! i installed asterisk on my computer, wrote down a few (hopefully correct!) lines in my sip.conf and then i ran asterisk. on the same system i ran twinkle to have some tries. however twinkle tells me "failed to create udp socket on 5060 ... address already in use". may i have this kind of attempts (i mean asterisk and twinkle on the same machine)? i'm afraid i can't... any hints? |
| 19:26.51 | kombi | no nat, dedicated ip, everything else went well so far. |
| 19:26.53 | *** part/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 19:27.41 | Mercestes | kombi: Hrm....try a debug 1 and sip debug and see if it gives you anything useful as far as an error or something. |
| 19:27.49 | Mercestes | kombi: And try allow=gsm in the sip.conf entries as well. |
| 19:29.41 | *** join/#asterisk anthm (n=anthm@000-457-779.area4.spcsdns.net) |
| 19:29.41 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:29.44 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:30.51 | falz | cheshair: you already have something listening on udp 5060, such as asterisk is already running. |
| 19:31.10 | Innatech | b1shop: http://www.smarthome.com/30532.html "proliphix IP thermostat" ....looks like an IP addressable thermostat that will do what you want. |
| 19:31.32 | lesouvage | I have a b410p card and for now its only purpose is to provide a hardware timer for Asterisk. I did a succesfull make b410p. How can I check if Asterisk actually is using the card as its timesource. |
| 19:31.52 | Mercestes | lesouvage, try to join a conference. |
| 19:32.04 | Innatech | b1shop: smarthome mostly sells X10 equipment, so I'd try and check it out before ordering the thing to make sure it's worthwhile. |
| 19:32.25 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 19:32.27 | lesouvage | mercestes: thanks, I should have thought of this myselve. |
| 19:32.50 | cheshair | falz: yeah i was afraid i was doing something dirty... :-) any ideas to have some tries? should i better emulate a second os with qemu? |
| 19:33.02 | Mercestes | Just be sure ztdummy is not loaded as a module, and there ya go. ;) |
| 19:33.18 | Mercestes | cheshair, Um, I would suggest changing the port on one or the other. |
| 19:33.38 | falz | netstat -anp | grep 5060 |
| 19:33.55 | _VoiceMeUp_COM | how bout , hmm |
| 19:33.58 | kombi | Mercestes: was that for me? |
| 19:34.02 | _VoiceMeUp_COM | sip is udp right ? |
| 19:34.03 | falz | will tell you what process is listening on things with '5060' in it |
| 19:34.06 | _VoiceMeUp_COM | signaling ? |
| 19:34.16 | _VoiceMeUp_COM | so its the sip .stack that i need to look at ? |
| 19:34.57 | Innatech | So, there's a little ISP in LA here , advertising a 1.5 T1 for $175/mo. Does that seem crazy to anyone else? I was just about to decide on an SDSL provider when I saw that. Seems fishy, no? |
| 19:34.57 | kombi | Mercestes: I'll re-compile zaptel with everything unselected but ztdummy.. |
| 19:35.05 | Wag3Slav3 | I can't connect to numbers that don't give at least one ring. Can some one help me? Sip trunk to sip phone. |
| 19:35.19 | _VoiceMeUp_COM | whis isp ;) |
| 19:35.24 | alrs | Innatech: You think it's a real T1, or "t1 class"? |
| 19:35.28 | Mercestes | kombi: no |
| 19:35.30 | falz | does'nt seem crazy to me. my company gets local loops from telcos for about $100, if we were crazy that extra $75 would be profit if we wanted to give unlimited bandwidth |
| 19:35.32 | aptura | Innatech unless thay want to live on razor thin margins its hard to say |
| 19:35.34 | Mercestes | kombi: Don't do that. |
| 19:35.43 | alrs | Innatech" There are some fixed-wireless guys in LA that call their service "T" |
| 19:36.04 | Innatech | alrs: they're saying its a real T1. The equipment sounds right. It's not that fixed wireless stuff, I've seent that around. |
| 19:36.12 | cheshair | Mercestes: is it so easy?? shouldn't they listen on the same port? you mean they don't use 5060 to talk together? (cheshair is a little bit stupid... :-( ) |
| 19:36.45 | Innatech | alrs: http://www.am6.net/services/biz/t1.html |
| 19:36.50 | kombi | Mercestes: rather everything on? no digium hardware in this box.. |
| 19:37.04 | Mercestes | kombi: ... |
| 19:37.13 | falz | that's a real t1, they're talking about the local loop and stuff |
| 19:37.13 | Supaplex | has anyone got a verizon razr to use voicemail attachments from asterisk? file(1) says they're both GSM 6.10 encoded wav files. |
| 19:37.20 | Mercestes | kombi: You should *know* what to compile in and what you should not because you are familiar with what hardware exists in your box. |
| 19:37.42 | Mercestes | Kombi: The answer is, compile in what you *need* and nothing you do not *need* if you have no zaptel hardware then you only need ztdummy |
| 19:37.42 | Innatech | yeah. Seems crazy cheap....but they're close enough that I can drive over there and scream at them if anything bad happens. Tempting. |
| 19:37.59 | kombi | Mercestes: I'll try that now.. |
| 19:38.11 | Mercestes | cheshair: What does twinkle do? |
| 19:38.14 | alrs | Innatech: I'm in Echo Park, I could be there on my bicycle machine in 10 minutes |
| 19:38.34 | Innatech | alrs: heh, and you've never heard of em, eh? Hmm. |
| 19:38.59 | Toerkeium | hello ! |
| 19:39.05 | cheshair | Mercestes: twinkle is a softphone like ekiga and refuses to start cause it can't use 5060 |
| 19:39.10 | alrs | Innatech: I've never paid for my own T1, I just shop them every so often |
| 19:39.10 | Innatech | Still, I'd rather pay $175 for a managed 1.5 t1 than $150 for a 768 SDSL. |
| 19:39.21 | Toerkeium | does anyone works with ViciDial? |
| 19:39.23 | Innatech | alrs: ah, OK. |
| 19:39.23 | *** part/#asterisk Wag3Slav3 (n=gregg@71-32-119-21.bsmr.qwest.net) |
| 19:39.48 | Mercestes | cheshair, That doesn't sound right. |
| 19:40.02 | alrs | Innatech: I have a feeling that $175 price is with a three-year contract |
| 19:40.19 | Mercestes | cheshair, One should be *listening* to 5060 and the other should be authenticating to port 5060, but..then again, I've never tried to run a softphone on my asterisk box |
| 19:40.44 | Innatech | alrs: might very well be. Still, that beats the pants off the competing prices I've seen even w/contracts. |
| 19:41.52 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
| 19:41.53 | cheshair | Mercestes: how do you have some tests? i have only one computer at the moment... |
| 19:42.57 | cheshair | Mercestes: maybe i misconfigured asterisk so that now it is using 5060 both for listening and authenticating? |
| 19:43.39 | kombi | Mercestes: can you knock calls off a conference from CLI? |
| 19:43.49 | [TK]D-Fender | cheshair, You can't have a softphone running on 5060 on the same box as *. They'd fight for listening control over the SIP port. |
| 19:43.53 | Mercestes | <PROTECTED> |
| 19:44.21 | [TK]D-Fender | cheshair, You'd have to configure your softphone to use another port like 5061 and set your sip entry to expect that port. |
| 19:46.16 | cheshair | [TK]D-Fender: i see, i'll fix immediately my stupid mistake! thank you!! and thanks to Mercestes too! |
| 19:46.28 | Mercestes | np |
| 19:47.25 | [TK]D-Fender | cheshair, No need to think of it as being stupid and you should try to stop talking aboutself like that. People may begin to believe you ;) |
| 19:48.43 | cheshair | [TK]D-Fender: no you're wrong... you all guys know i'm so smart! :-) |
| 19:48.51 | *** join/#asterisk anthm][ (n=anthm@000-457-779.area4.spcsdns.net) |
| 19:49.03 | [TK]D-Fender | cheshair, Oh yeah, and false bravado will get you caught INSTANTLY. |
| 19:51.24 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 19:51.26 | *** part/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 19:51.34 | kombi | hmm, weiredness continues, as soon a meetme conference is started, Internal RTCP NTP clock skew detected - messages fly in |
| 19:51.45 | cheshair | [TK]D-Fender: kidding aside i'm using * since only few hours, so i'm just learning the very first steps. luckily i found a great deal of help here |
| 19:51.50 | *** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net) |
| 19:51.54 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 19:52.04 | *** part/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 19:52.18 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 19:52.24 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:53.28 | Mercestes | Kombi: Yea, that would be the 1.4.5 shittiness I was referring to earlier that I've seen on other systems. |
| 19:53.41 | Mercestes | Kombi: I would unapologeticlly bug report it and hope you get an answer. |
| 19:54.01 | Mercestes | or a reference to another bug report. |
| 19:54.16 | Mercestes | google turned up nothing for me.. |
| 19:54.19 | kombi | hmm, really? apart from this that release seems ok.. |
| 19:54.33 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 19:54.39 | _VoiceMeUp_COM | ahah |
| 19:55.00 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 19:55.10 | *** join/#asterisk bkruse_home (n=kruz@50.49.146.64.transedge.com) |
| 19:55.51 | *** join/#asterisk vader-- (n=me@204.183.88.101) |
| 19:55.52 | vader-- | hello |
| 19:55.55 | Mercestes | kombi: I agree with your statement. |
| 19:56.09 | Qwell[] | Do you guys have anything strange with your setups? |
| 19:56.15 | vader-- | has anyone seen an issue where a ZAP channel will not recognize when a phone is picked up |
| 19:56.17 | Qwell[] | say, in vmware or xen or something silly? |
| 19:56.25 | vader-- | we have a digium TDM 2400 card |
| 19:56.39 | vader-- | and on thursday our fax machine was able to send faxes and receive them |
| 19:56.40 | Mercestes | Qwell[]: on my end it should be RHEL on a Dell PE server. |
| 19:56.48 | Qwell[] | Should be? O.o |
| 19:56.49 | vader-- | on friday it stopped sending |
| 19:56.50 | Mercestes | Qwell[]: Old co-workers of mine. |
| 19:57.01 | vader-- | now with a regular telephone attached it doesn't have a dial tone |
| 19:57.07 | vader-- | but when you call the line it rings |
| 19:57.12 | Mercestes | Qwell[]: I left their service som etime ago but I'm pretty familiar with their setup. They wouldn't know to xen or Vmware it. |
| 19:57.22 | vader-- | when im in the asterisk console it doesn't show me picking the phone up |
| 19:57.30 | vader-- | you guys know of anything i can try? |
| 19:57.36 | stoffell_w | Qwell, i'm planning on doing some testing with xensource and asterisk.. |
| 19:57.37 | Qwell[] | vader--: sounds like your card isn't configured, or zaptel isn't loaded or something |
| 19:57.46 | vader-- | it was working fine for a year now |
| 19:57.50 | kombi | Mercestes: how much of a headache would it be to roll back to 1.2? |
| 19:57.53 | vader-- | and all our other zap channels are fine |
| 19:58.00 | vader-- | no one changed anything on the telephone config |
| 19:58.10 | Mercestes | kombi: emerge -Ca asterisk and then mask > 1.2 and emerge asterisk |
| 19:58.16 | Mercestes | Kombi: Oh yea, your probably not using Gentoo. |
| 19:58.19 | Qwell[] | vader--: it's possible that the module for the tdm2400p isn't being loaded |
| 19:58.45 | Mercestes | Kombi: Umm, something like a cd /usr/src/asterisk && make clean && make distclean and a nice cd .. and a rm-dvfr ./asterisk-1.4.shitty and then download and compil enew source. |
| 19:58.59 | Mercestes | kombi: Are you using Xen or Vmware on your box? |
| 19:59.04 | kombi | Merceses: ;) nay, not any kind of pre packaging |
| 19:59.19 | kombi | no Xen or Vmware |
| 19:59.29 | tzafrir_home | make clean && make distclean? |
| 19:59.31 | Qwell[] | only reason I ask is because those are notoriously bad a timing |
| 19:59.34 | Mercestes | kombi: Qwell perked his ears btw |
| 19:59.41 | tzafrir_home | nice. Considering distclean calls clean first |
| 19:59.50 | Qwell[] | and that you're removing the dir anyways :p |
| 19:59.54 | Mercestes | tzafrir_home, I like to be very clean. |
| 19:59.55 | cheshair | [TK]D-Fender: hmmm i didn't manage to fix the problem: i wrote "bindport=5061" in my sip.conf and restarted *, then i told twinkle to use 5061. now it complains "5061 is already in use"... |
| 20:00.02 | Mercestes | and thorough |
| 20:00.08 | kombi | Mercestes: I just fear to loose all the stuff that took me hours to fix..;) |
| 20:00.15 | [TK]D-Fender | cheshair, NO! Again you are making them fight over the SAME PORT! |
| 20:00.19 | Mercestes | kombi: Is it under /usr/src/asterisk? |
| 20:00.27 | kombi | yip.. |
| 20:00.35 | [TK]D-Fender | cheshair, * should bind to 5060, your softphone to 5061! |
| 20:00.39 | Mercestes | [TK]D-Fender, ROFLMAO |
| 20:00.39 | tzafrir_home | Mercestes, make mrproper, then |
| 20:00.45 | kombi | guess I just roll a new one for the heck of it |
| 20:01.02 | kombi | with make mrproper |
| 20:01.03 | cheshair | [TK]D-Fender: i see... but then how can my twinkle to talk to my *? |
| 20:01.09 | [TK]D-Fender | cheshair, in that PHONE'S entry in sip.conf add "port=5061" |
| 20:01.19 | Mercestes | tzafrir_home: You are talking to the creator of "emerge -Duanetv world" |
| 20:01.29 | [TK]D-Fender | cheshair, You change what * is expecting for the PHONE, not what * expects to bind to normally. |
| 20:01.41 | _VoiceMeUp_COM | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1320.html |
| 20:01.43 | cheshair | [TK]D-Fender: ahhh that's the trick... |
| 20:01.50 | *** part/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 20:01.52 | [TK]D-Fender | cheshair, Port 5061 can CALL 5060. |
| 20:01.52 | _VoiceMeUp_COM | can i use mac in zapt ? |
| 20:01.57 | *** part/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 20:01.59 | [TK]D-Fender | cheshair, Try again. |
| 20:02.05 | cheshair | [TK]D-Fender: immediately |
| 20:02.15 | Mercestes | _VoiceMeUp_COM, There are no macintosh builds for zapt. |
| 20:02.35 | tzafrir_home | whatever happened to it? |
| 20:02.41 | _VoiceMeUp_COM | i mean mac addy lol |
| 20:02.58 | _VoiceMeUp_COM | basicaly trying to make a zap group use a specific ip |
| 20:03.04 | tzafrir_home | isn't there even a mac ztdummy? |
| 20:03.13 | _VoiceMeUp_COM | ao zap/g1 uses 1.2.3.4 and zap/g2 uses 54.65.7.8 |
| 20:03.22 | sergee | are there any win32 gui to edit asterisk configs? |
| 20:03.42 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 20:04.00 | _VoiceMeUp_COM | but aint my problem a flaw in the code ? |
| 20:04.21 | _VoiceMeUp_COM | if reposnding to a sip inqueiry it should reply on the ip it was received from.. .anyways |
| 20:06.03 | *** join/#asterisk mrdigital (n=mrdigita@207-172-229-100.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 20:06.09 | cheshair | [TK]D-Fender: ok! that did the work! :-) however now twinkle tells me "user, registration failed: 404 Not Found"... i'll try to fix it and i'll let you know what's happening |
| 20:07.22 | [TK]D-Fender | cheshair, How they're talking and at least disagreeing... good start. |
| 20:07.29 | [TK]D-Fender | cheshair, Now fix your user/pass scenario |
| 20:07.58 | cheshair | [TK]D-Fender: sure, i'm working on it |
| 20:09.59 | *** join/#asterisk soylentgreen (n=fgast@bb1-fe0.only640k.org) |
| 20:10.38 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 20:10.39 | vader-- | tdk do you have any ideas as to y im seeing this issue |
| 20:11.09 | *** part/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 20:11.14 | tzafrir_home | sergee, putty :-( |
| 20:11.33 | tzafrir_home | winscp is a slightly friendlier one |
| 20:12.06 | _VoiceMeUp_COM | winscp sucks |
| 20:12.10 | _VoiceMeUp_COM | sftpdrive.com |
| 20:12.14 | _VoiceMeUp_COM | use that |
| 20:12.19 | _VoiceMeUp_COM | winscp doesn use keys |
| 20:12.25 | _VoiceMeUp_COM | well bad ly it does |
| 20:12.44 | _VoiceMeUp_COM | plus winscp freezes as it looks like a java app |
| 20:12.57 | robl^ | coreftp is ok |
| 20:12.57 | tzafrir_home | filezilla also supports sftp . IIRC so does total commander (previously: wincommander) |
| 20:13.44 | tzafrir_home | "_VoiceMeUp_COM> plus winscp freezes as it looks like a java app": care to explain? |
| 20:13.48 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.144.42) |
| 20:15.20 | *** join/#asterisk gardo (n=gardo@121.97.196.53) |
| 20:15.24 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:16.43 | sergee | tzafrir_home: i have a crazy idea, to build a kind of windows package: asterisk + asterisk-addons + some manager gui + some configuration gui :) |
| 20:17.08 | sergee | tzafrir_home: compiling 1.4.5 right now :) |
| 20:18.09 | Mercestes | sergee: Please, asterisk has enough problems without windows mucking up the waters. |
| 20:18.35 | cheshair | [TK]D-Fender: step by step things get better and better! :-) "user, registration succeeded (expires = 3600 seconds)" what comes next? at the moment i can't make calls: "Line 1: call failed. 404 Not Found" i'll work on it immediately |
| 20:19.07 | ManxPower | cheshair: that message is ALWAYS one of 2 issues. |
| 20:19.10 | obnauticus | Has anyone here setup Festival? |
| 20:19.12 | [TK]D-Fender | cheshair, Now you need to make sure your sip device is pointed to an appropriate context in your dialplan |
| 20:19.25 | obnauticus | festival with asterisk* |
| 20:19.26 | ManxPower | 1) the incoming call not matching a sip.conf section |
| 20:19.33 | [TK]D-Fender | ^^^ |
| 20:19.53 | ManxPower | 2) the context= line does not point to a context with a matching exten => line for the digits you are dialing. |
| 20:19.59 | [TK]D-Fender | ^^^ |
| 20:20.32 | cheshair | ManxPower & [TK]D-Fender: i see... |
| 20:22.01 | _VoiceMeUp_COM | waut |
| 20:22.01 | sergee | Mercestes: oh my :) i don't force you to use it, right? |
| 20:22.01 | ManxPower | an easy way to test this is put context=INVALID in sip.conf [general], put the correct context= line for each of the other sip.conf entries, then put in extensions.conf [INVALID] and exten => _.,1,Noop(Unauthenticated call to ${EXTEN}) |
| 20:22.01 | _VoiceMeUp_COM | wait !!! cant i use domain= IP to use in sip.conf ? |
| 20:22.01 | _VoiceMeUp_COM | in the peer def |
| 20:22.01 | _VoiceMeUp_COM | this way it could force it ? |
| 20:22.01 | sergee | Mercestes: there is even nothing to use yet :) |
| 20:22.08 | MindTheGap | do Gotoif() accepts OR? like: exten => *78*,2,Gotoif($[${code} =! "5" OR "21" OR "37"]?true:false) |
| 20:22.52 | ManxPower | MindTheGap: does "show application gotoif" say that it does? |
| 20:23.13 | ManxPower | or specifically README.variables with regards to the $[] contruct? |
| 20:24.34 | MindTheGap | ManxPower, no, it doesnt... didnt check README.variables though... |
| 20:25.01 | ManxPower | MindTheGap: It is really an evaluation issue, not a gotoif issue |
| 20:25.05 | ManxPower | perhaps this will help: http://www.fnords.org/~eric/macros.inc |
| 20:25.14 | _VoiceMeUp_COM | fromdomain i mean |
| 20:25.15 | _VoiceMeUp_COM | ill try |
| 20:25.16 | _VoiceMeUp_COM | i guess |
| 20:25.27 | ManxPower | exten => _XXXX,8,GotoIf($["${DIALSTATUS}" = "BUSY" | "${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "CONGESTION"]?12:9) |
| 20:25.37 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 20:25.56 | cheshair | do you think this sip.conf could work? |
| 20:26.01 | cheshair | http://pastebin.com/936267 |
| 20:26.53 | ManxPower | [TK]D-Fender: I think today is another "ignore everyone's advice day" today. |
| 20:27.03 | ManxPower | cheshair: I'm sorry I cannot help you further |
| 20:27.24 | MindTheGap | ManxPower, thanks... |
| 20:27.41 | cheshair | ManxPower: doh i guess i did something wrong! sorry!! i'll try to fix it immediately |
| 20:29.02 | [TK]D-Fender | cheshair, Check. Your. DIALPLAN. |
| 20:29.55 | ManxPower | cheshair: what you didn't do is follow my advice |
| 20:30.15 | ManxPower | cheshair: exten => _XXXX,8,GotoIf($["${DIALSTATUS}" = "BUSY" | "${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "CONGESTION"]?12:9) |
| 20:30.17 | ManxPower | argh! |
| 20:30.23 | cheshair | sorry guys, the point is i can't always see what you exactly mean with your words |
| 20:30.41 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 20:30.42 | cheshair | eg: what's my dialplan? sip.conf? |
| 20:30.42 | kombi | before I venture into this: to install 1.2 over 1.4: ./configure make mrproper make make install? |
| 20:30.43 | ManxPower | cheshair: I will show you |
| 20:31.05 | ManxPower | cheshair: http://pastebin.com/936272 |
| 20:31.08 | tzafrir_home | kombi, there's no 'make mrproper' |
| 20:31.12 | cheshair | ManxPower: i'll try to listen as carefully as i can |
| 20:31.18 | tzafrir_home | sorry for misleading you |
| 20:31.28 | kombi | tzafrir_home: I believed you!! |
| 20:31.55 | kombi | ;), make clean, make real clean make dist clean |
| 20:32.37 | kombi | make terrify clean make petrify clean make stupify clean? |
| 20:33.16 | kombi | trafrir_home: no, seriously, will I shoot myself into open space or might this work? (just so I know..) |
| 20:33.38 | cheshair | ManxPower: was i supposed to write all that stuff on my own?! :-) i'll read it carefully, then i'll ask you st. |
| 20:34.00 | tzafrir_home | kombi, there's 'clean' and 'dist-clean' |
| 20:34.07 | ManxPower | kombi: MANY people downgrade just fine. In fact make install will even tell you what modules that are installed that are not built by 1.2 |
| 20:34.26 | ManxPower | cheshair: if you read my advice, then look at the pastebin, you will see |
| 20:34.28 | kombi | I'll dare doing it hten.. |
| 20:34.30 | kombi | then.. |
| 20:34.40 | *** join/#asterisk hi365_m (i=HydraIRC@212.199.22.3.forward.012.net.il) |
| 20:34.43 | ManxPower | kombi: I assume you are not an a production box. |
| 20:34.49 | DragoraN | is it right to set lowdelay on RTP and highreliability on SIP signaling? |
| 20:34.56 | tzafrir_home | consider also saving a backup copy of /etc/asterisk and/or /usr/lib/asterisk/moduels |
| 20:34.59 | hi365_m | anyone here with a sangoma a102d? |
| 20:35.08 | ManxPower | DragoraN: it won't matter at all once your call goes across the internet |
| 20:35.17 | hi365_m | hey tzafrir |
| 20:35.23 | ManxPower | hi365_m: 2-port Sangome T-1 card, right? |
| 20:35.26 | hi365_m | yup |
| 20:35.28 | DragoraN | ManxPower: ISP must support it? |
| 20:35.36 | ManxPower | DragoraN: no ISPs support it |
| 20:35.43 | DragoraN | ManxPower: so? |
| 20:35.57 | DragoraN | oh |
| 20:35.58 | lesouvage | I have a b410p card installed succesfully. Meetme is working but this is in the cli "Unable to open '/dev/zap/pseudo': No such device or address" and before entering the conference I hear "this is not a valid conference number" How can I fix this? |
| 20:36.00 | ManxPower | DragoraN: so it does not matter what you set those options to. |
| 20:36.06 | kombi | haaa.. no configure script in 1.2? really? |
| 20:36.13 | hi365_m | ManxPower: yup. have you got one? |
| 20:36.14 | ManxPower | kombi: correct |
| 20:36.23 | ManxPower | hi365_m: yes, I have them |
| 20:36.37 | tzafrir_home | lesouvage, get zaptel/ztdummy |
| 20:36.55 | ManxPower | tzafrir_home: why? He has a real Digium card. |
| 20:37.05 | kombi | ..and no make menuselect.. |
| 20:37.24 | Qwell[] | ManxPower: Those cards don't use zaptel |
| 20:37.25 | tzafrir_home | ManxPower, people here will start believing you |
| 20:37.34 | mocker | I hate fax machines. |
| 20:37.39 | ManxPower | kombi: make clean && make install |
| 20:37.48 | DragoraN | ManxPower: so how to guarentee site-to-site voip? |
| 20:38.21 | DragoraN | ManxPower: ISDN? |
| 20:38.21 | mocker | ~ fax |
| 20:38.21 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
| 20:38.24 | ManxPower | DragoraN: you put in your own T-1/E-1/whatever between the two sites and you manage all traffic on those links with routers that support QoS |
| 20:38.35 | *** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net) |
| 20:38.48 | ManxPower | You cannot do QoS over the internet. |
| 20:38.56 | hi365_m | ManxPower: PM? |
| 20:39.00 | ManxPower | Not in any sense of the real world. |
| 20:39.00 | lesouvage | Qwell: I installed mIsdn with make b410 . Asterisk seem to look for ztdummy before it discovers the b410p. |
| 20:39.06 | DragoraN | ManxPower: ok, thx for explain |
| 20:39.13 | ManxPower | hi365_m: if you want private consulting then you'll pay. If you want public help then ask on the channel |
| 20:39.16 | *** join/#asterisk eatmypiano (n=eatmypia@host86-132-181-229.range86-132.btcentralplus.com) |
| 20:39.33 | kombi | '/usr/bin/ld: cannot find -lssl <-- what is that? |
| 20:39.40 | hi365_m | no prob. was just gett a headace from all the post |
| 20:39.42 | Qwell[] | kombi: install openssl-devel |
| 20:39.46 | mocker | kombi: I'd guess SSL. |
| 20:39.47 | mocker | :) |
| 20:39.48 | kombi | thanks! |
| 20:39.55 | hi365_m | ManxPower: what is yout zttest results? |
| 20:40.04 | ManxPower | hi365_m: I have no idea. |
| 20:40.12 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 20:40.30 | *** join/#asterisk sci_05 (n=peter@waterfall.bestserversllc.net) |
| 20:40.38 | hi365_m | ManxPower: can i bother you to run a test? |
| 20:40.45 | ManxPower | --- Results after 7 passes --- |
| 20:40.46 | ManxPower | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.996512 |
| 20:40.53 | tzafrir_home | hi365_m, what do you in yours? |
| 20:40.53 | ManxPower | this is on a live system |
| 20:41.02 | hi365_m | 99-97 |
| 20:41.04 | hi365_m | :( |
| 20:41.39 | hi365_m | and alot of choppy audio :{ |
| 20:42.11 | ManxPower | hi365_m: do all the same troubleshooting options for Sangoma as for Digium |
| 20:42.40 | ManxPower | check for IRQ conflicts, other things that can cause this is running in graphics mode, onboard ethernet, onboard SATA, etc. |
| 20:43.13 | hi365_m | thats what ive been doing. irq seems fine. no graphics. |
| 20:43.14 | ManxPower | also check your timing |
| 20:43.20 | stoffell_w | hi365_m, or even iLO on a HP server can cause this ;) |
| 20:43.50 | ManxPower | hi365_m: SOMETHING in your system is locking interrupts for too long |
| 20:43.56 | Mercestes | hi365_m, or giv eSangoma a call. |
| 20:44.21 | hi365_m | sata would cause such issues? |
| 20:45.01 | stoffell_w | hi365_m, maybe the sata controller.. is it in raid? |
| 20:45.07 | hi365_m | nope |
| 20:45.22 | stoffell_w | what type of controller is it? |
| 20:45.23 | ManxPower | hi365_m: anything that locks interrupts for a long will cause this issue. SATA interfaces commonly lock interrupts for a very long time to give better performance at the expense of all other interrupt driven devices on the system |
| 20:45.37 | hi365_m | how can i test if its the sata? |
| 20:45.40 | ManxPower | Same goes for onboard ethernet |
| 20:45.59 | stoffell_w | ManxPower, leading to the conclusion that it's better to use SCSI disks? right? |
| 20:45.59 | hi365_m | stoffell_w: its onboard (intel server board) |
| 20:46.07 | ManxPower | hi365_m: you cannot "test" this. you disable the device and see of it fixes the problem |
| 20:46.25 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
| 20:46.38 | hi365_m | if i dissable sata = no hard drives |
| 20:46.42 | ManxPower | stoffell_w: I've not heard of people with SCSI having these issues, but there is NOTHING about SCSI that would prevent this problem |
| 20:46.45 | mocker | Anyone used the Attractel t.38 fax stuff? |
| 20:46.50 | ManxPower | hi365_m: then get a PCI SATA card |
| 20:47.15 | hi365_m | problem is its a 1u server, allready taken by the sangoma card |
| 20:47.25 | mocker | ~ t38 |
| 20:47.25 | jbot | it has been said that t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 20:47.27 | ManxPower | SCSI does tend to load the host processor less that IDE style things, but the driver could still lock interrupts |
| 20:47.29 | stoffell_w | hi365_m, try with an IDE disk maybe, just to see if it works.. |
| 20:47.31 | [TK]D-Fender | hi365_m, SATA is fine, SCSI is fine. Only Digium cards have problems with these. |
| 20:47.31 | ManxPower | hi365_m: it sucks to be you. |
| 20:47.51 | ManxPower | [TK]D-Fender: this guy has a Sangoma with poor zttest results |
| 20:48.02 | holiday_42 | hm. use livecd? |
| 20:48.19 | J4k3 | USB flash! |
| 20:48.25 | J4k3 | ;) |
| 20:48.25 | holiday_42 | better! |
| 20:48.26 | stoffell_w | use livecd, holiday_42 has a good point :-) |
| 20:48.43 | J4k3 | but it tends to be on the horribly offensive side |
| 20:48.47 | J4k3 | when it comes to I/O overhead |
| 20:48.59 | ManxPower | and we both know that the same basic issues affect Sangoma as well as Digium. It's just that Sangoma seems to do less work in it's ISR and so the IRQ latency has to be much worse for Sangoma to have a problem |
| 20:49.00 | J4k3 | most CF cards do mode 2 or somesuch... dirty and slow |
| 20:49.27 | ManxPower | hi365_m: You really need to call Sangoma tech support |
| 20:49.30 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 20:49.31 | J4k3 | and remember folks, you can always soft-raid flash devices ;) |
| 20:49.47 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 20:50.17 | sci_05 | nayone got any ideas as to what would cause a call to go silet for a few seconds when in the middle of a call? I am using iax thru a vtun |
| 20:50.37 | hi365_m | i guess well do that tommorow |
| 20:50.40 | *** part/#asterisk ManxPower (n=manxpowe@42.sub-75-200-20.myvzw.com) |
| 20:51.20 | syle | raid flash devices, what would be point in that |
| 20:51.24 | Innatech | j4k3: OOC, have you done much with softraid on CF? I'm thinking about going that route for some router boxes I need to build. |
| 20:51.28 | stoffell_w | hi365_m,or put the card in another box.. i've seen a cheap pc perform better then some 'server' mainboards.. |
| 20:51.29 | J4k3 | they'll make nice bedroom phones. |
| 20:51.44 | J4k3 | Innatech: yep, I've done it with as-expected results. |
| 20:51.46 | syle | anyone good cordless sip phones yet |
| 20:51.59 | Innatech | J4k3: cool. thanks. |
| 20:52.01 | J4k3 | Innatech: you're best off leaving the FS in read-only mode if possible. |
| 20:52.15 | Innatech | j4k3: yes, I figured. ;) |
| 20:52.23 | cheshair | ManxPower: hm, i didn't succeed with last sip.conf modifications... i just used your http://pastebin.com/936272 ... i'm still told 404 not found by twinkle... maybe i don't use the proper syntax when i make the call? |
| 20:52.32 | J4k3 | yeah... I have managed to kill a CF card in 2 weeks with writes :) |
| 20:53.13 | *** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net) |
| 20:53.16 | Innatech | heh. Yeah, l can see that happening w/o much effort. |
| 20:53.40 | [TK]D-Fender | cheshair, NEVER use pastebin.com again ok. Its seriously slow and screwed up. use pastebin.ca from now on. |
| 20:53.47 | obnauticus | [TK]D-Fender can i ask a quick question? |
| 20:53.52 | obnauticus | well |
| 20:53.54 | obnauticus | i'll ask it anyway |
| 20:53.55 | obnauticus | lol. |
| 20:54.03 | cheshair | [TK]D-Fender: sure |
| 20:54.04 | Supaplex | cheshair: or http://rafb.net/paste |
| 20:54.05 | [TK]D-Fender | obnauticus, You learn quick Padawan |
| 20:54.10 | obnauticus | how do i have caller defined variables |
| 20:54.16 | *** part/#asterisk hi365_m (i=HydraIRC@212.199.22.3.forward.012.net.il) |
| 20:54.17 | obnauticus | [TK]D-Fender i usually don't ask to ask |
| 20:54.20 | obnauticus | no idea why i did that. |
| 20:54.30 | cheshair | ok, thanx Supaplex |
| 20:54.33 | *** join/#asterisk SuD (n=Ask@89.140.32.2.static.user.ono.com) |
| 20:54.42 | [TK]D-Fender | obnauticus, please clarify what you mean by "caller defined variables" |
| 20:54.46 | obnauticus | like |
| 20:54.52 | obnauticus | "Dial the number of your voicemail" |
| 20:55.30 | Innatech | so, DTMF input. |
| 20:55.37 | obnauticus | ya ya |
| 20:55.40 | [TK]D-Fender | obnauticus, rewind your entire idea, it doesn't make sense. |
| 20:56.08 | Innatech | He wants to react to arbitrary strings of DTMF tones. |
| 20:56.08 | obnauticus | [TK]D-Fender clients can call in and check their voicemail |
| 20:56.30 | [TK]D-Fender | obnauticus, is SOUNDS like English, but like Chris Rock said in "Rush Hour" : "Man ain't NOBODY understand the words comin' outta yo mouth!" |
| 20:56.43 | obnauticus | I'm trying to make it simple. |
| 20:56.49 | obnauticus | for myself lol. |
| 20:57.01 | [TK]D-Fender | obnauticus, You want them to be able to pick up VM remotely from within an IVR you already made? |
| 20:57.11 | obnauticus | ya |
| 20:57.28 | [TK]D-Fender | obnauticus, Exten => 12345,1,VoiceMailMain() <- There |
| 20:57.41 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-70-107-173-57.ny325.east.verizon.net) |
| 20:57.48 | [TK]D-Fender | obnauticus, Just shove that in your IVR and BAM people hit 123 and can pick up voicemail. |
| 20:57.50 | mocker | Anyone have a decent fix for faxes showing up corrupted sometimes? |
| 20:57.55 | mosty | is there a way to extract a call summary from the asterisk full log, so that i can see how each call was terminated? i have pbx that appears to drop everybodies calls all at once, and i'm trying to figure out why. there are too many calls happening at once to just watch the console |
| 20:57.55 | mocker | Using rxfax. |
| 20:58.11 | [TK]D-Fender | mocker, Stop using SpanDSM and use a real machine and an entirely seperate line. |
| 20:58.55 | stoffell_w | mocker, or use a digium IAXy or even iaxmodem.. everything is better then rxfax :-) |
| 20:59.40 | tzafrir_home | mocker, which version of spandsp? |
| 21:00.21 | *** join/#asterisk Rusty1 (n=Rusty1@cpe-72-226-96-74.nycap.res.rr.com) |
| 21:00.28 | tzafrir_home | coppice said that version 0.04 should be better than previous ones |
| 21:01.43 | mocker | tzafrir_home: Not sure.. |
| 21:01.48 | kombi | hrmpf, made and installed 1.2, ditto addons, ztdummy still loaded, no meetme registered.. |
| 21:01.51 | mocker | tzafrir_home: I'll have to dig around. |
| 21:02.07 | tzafrir_home | I haven't tested it myself |
| 21:02.51 | mocker | [TK]D-Fender: That solution stinks. :( |
| 21:03.05 | mocker | But rxfax does seem to suck./ |
| 21:03.25 | stoffell_w | psst mocker... iaxmodem.. |
| 21:04.46 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 21:05.17 | sandorp | can someone recommend a good/cheap IP phone that works well with Asterisk? |
| 21:05.35 | [TK]D-Fender | sandorp, Polycom IP 320 = $95USD |
| 21:05.46 | cheshair | guys i'm keeping on receiving "call failed 404 not found"... i just write on my twinkle call command line: 611 or 100 or sip:611@192.168.1.3 or st. similar... here comes my sip.conf: http://pastebin.ca/589148 (powered by ManxPower :-)) |
| 21:05.57 | mocker | stoffell_w: Really works that much better? |
| 21:06.16 | DragoraN | when asterisk fires Dial command, caller is charged? |
| 21:06.19 | sandorp | [TK]D-Defender: thanks |
| 21:06.32 | stoffell_w | mocker, the IAXy is preferred but I've switched several boxes from rxfax to iaxmodem (with hylafax backend), much better! |
| 21:06.47 | mosty | i have a bunch of log messages "app_dial.c: Unable to forward voice", is there any way to figure out why asterisk was unable to do that? |
| 21:06.52 | kombi | the order in which to install: *, zaptel, libpri, addons <-- correct? |
| 21:07.02 | [TK]D-Fender | cheshair, enable sip debug in * CLI and pastebin the failed attempt |
| 21:07.09 | sandorp | kombi: thats the way I do it |
| 21:07.10 | [TK]D-Fender | kombi, NO. |
| 21:07.12 | DragoraN | when asterisk fires Dial command and caller waits for pickup, caller is charged? |
| 21:07.16 | sandorp | actually, libe is first |
| 21:07.21 | [TK]D-Fender | kombi, libpri, zaptel, *, then addons. |
| 21:07.27 | kombi | thanks! |
| 21:08.18 | [TK]D-Fender | DragoraN, that is NOT a complete description, and where does "charge" come in? What tech is the caller using? What about the dial itself? What tech is THAT using? |
| 21:09.06 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 21:09.06 | DragoraN | [TK]D-Fender: i dont understand |
| 21:09.32 | [TK]D-Fender | DragoraN, What tesh is the person placing the Ccall using? |
| 21:09.46 | [TK]D-Fender | DragoraN, What tech is the person placing the call using? * |
| 21:09.48 | DragoraN | when does charging start for caller? (calling from landline to my voip-ed number) |
| 21:09.49 | obnauticus | [TK]D-Fender do you know of any web-based MeetMeAdmin stuff? |
| 21:09.52 | Qwell[] | [TK]D-Fender: I think your c jumped over a few chars... |
| 21:10.02 | obnauticus | because umm MeetMe administration is hard all through extensions and stuff |
| 21:10.03 | [TK]D-Fender | obnauticus, All the GUI's have their own. |
| 21:10.06 | DragoraN | and voiped number is asterisk |
| 21:10.42 | [TK]D-Fender | DragoraN, What kind of "line" is your "landline"? |
| 21:10.48 | DragoraN | normal PSTN |
| 21:10.53 | DragoraN | analog |
| 21:11.07 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 21:11.08 | [TK]D-Fender | DragoraN, Normal PSTN (with zaptel) considers the call answered as soon as it is PLACED. |
| 21:11.28 | [TK]D-Fender | DragoraN, If you want better progress detection, then get a digital link to the PSTN |
| 21:11.45 | DragoraN | [TK]D-Fender: yes I understand |
| 21:12.01 | DragoraN | i have SIP connection to PSTN at my ISP |
| 21:12.43 | DragoraN | when does my ISP consider incoming call towards me as placed? before ringing phones on my side or after it hangs up |
| 21:12.55 | [TK]D-Fender | DragoraN, Depends on THEM. |
| 21:13.00 | DragoraN | ahaa :) |
| 21:13.04 | DragoraN | thx |
| 21:13.15 | [TK]D-Fender | DragoraN, Place a call, watch the CLI output and you'll know when its considered "answered" |
| 21:13.25 | DragoraN | iam aware of asking this the PSTN provider at hotline ;) |
| 21:13.31 | lesouvage | What does * force to look for a pseudo device while there is a real and working card in the server? I have the b410 up and running, meetme is working, misdnportinfo looks ok to me, but having a message "thi is not a valid conference number is very annoying. Any suggestion? |
| 21:13.41 | [TK]D-Fender | DragoraN, Some "accept" your call right away, other pass on the PSTN call progress. |
| 21:14.14 | [TK]D-Fender | lesouvage, Check to see if your meetme.conf is actually SET UP RIGHT. |
| 21:14.19 | mosty | lesouvage, does meetme.conf define the conference? |
| 21:14.24 | [TK]D-Fender | lesouvage, And pastebin all of this proof BEFORE naggin us ;) |
| 21:15.21 | cheshair | [TK]D-Fender: these are the last lines, i hope they'll be enough http://pastebin.ca/589171 |
| 21:15.57 | [TK]D-Fender | # |
| 21:15.57 | [TK]D-Fender | Looking for 611 in internal (domain 192.168.1.3) |
| 21:15.57 | [TK]D-Fender | # |
| 21:15.57 | [TK]D-Fender | <PROTECTED> |
| 21:15.57 | [TK]D-Fender | # |
| 21:15.58 | [TK]D-Fender | <--- Reliably Transmitting (no NAT) to 192.168.1.3:5061 ---> |
| 21:16.00 | [TK]D-Fender | # |
| 21:16.02 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 21:16.06 | [TK]D-Fender | oops |
| 21:16.08 | [TK]D-Fender | dammit |
| 21:16.10 | [TK]D-Fender | Stupid XCHAT |
| 21:16.49 | [TK]D-Fender | cheshair, that tells yout aht your dialplan as loaded does not match |
| 21:17.09 | [TK]D-Fender | Looking for 611 in internal <----------------------- |
| 21:17.15 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 21:17.17 | robl^ | [TK]D-Fender: using Pidgin ;-) |
| 21:17.19 | FuriousGeorge | hey all |
| 21:17.42 | [TK]D-Fender | robl^, I tried Pidgin (as GAIM) and Esc key would EXIT the chat! |
| 21:17.50 | [TK]D-Fender | robl^, Pissed me right off! |
| 21:18.00 | robl^ | it does?!!? never did that to me |
| 21:18.00 | [TK]D-Fender | robl^, and I recall not having font control, etc. |
| 21:18.09 | [TK]D-Fender | robl^, Did to me. |
| 21:18.27 | [TK]D-Fender | robl^, In all fairness I will try again as I use Pidgin as my IM. |
| 21:18.47 | robl^ | fonts are normally part of IRC. it's a nonstandard add on. but pidgin has a plug-in that enables it |
| 21:19.00 | robl^ | er.. are NOT normally |
| 21:19.19 | cheshair | [TK]D-Fender: st wrong in http://pastebin.ca/589148 ? |
| 21:19.41 | [TK]D-Fender | "JUST four per cent of US adults are virgins, but a fifth have tried hard drugs such as cocaine and crack, a new study shows. What most alarms researchers is how young they start." _ Yaya for society! |
| 21:19.47 | *** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 21:20.01 | [TK]D-Fender | cheshair, And what exactly am I looking at? |
| 21:20.12 | cheshair | [TK]D-Fender: my sip.conf |
| 21:20.12 | DrRighteous | Anyone know if there is a way to set the ptime for sip in asterisk? |
| 21:20.17 | [TK]D-Fender | cheshair, ............. |
| 21:20.20 | [TK]D-Fender | cheshair, .................. |
| 21:20.21 | robl^ | I use hard drugs.. caffeine, sugar, caffeine, and caffeine |
| 21:20.32 | [TK]D-Fender | cheshair, the 2nd half is supposed to be EXTENSIONS.CONF!!!!!!!!!!!!!!!!!!!!! |
| 21:21.00 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:21.14 | [TK]D-Fender | cheshair, # |
| 21:21.14 | [TK]D-Fender | [internal] onward is your friggen DIALPLAN. Stop now and go back to reading the book. |
| 21:21.41 | *** join/#asterisk marchon (n=marchon@c-75-68-31-152.hsd1.nh.comcast.net) |
| 21:21.47 | Innatech | yikes.... |
| 21:21.56 | [TK]D-Fender | cheshair, If you don't know that "exten =>" clearly implies extensions.conf and is your dialplan you are going to induce ulcers in yourself and others. |
| 21:22.08 | [TK]D-Fender | </bile> |
| 21:22.16 | marchon | I need to get a timing board for my asterisk. I would like to have it shipped today. |
| 21:22.22 | [TK]D-Fender | ask;fdkl;fasjdfsdlggretgertiop[ugfsopdfumgopsidfumgiopsdumfgsdfg |
| 21:22.32 | _VoiceMeUp_COM | rtewwwwwwwwwwwwwwwwwwwwwwwwwwqe423 |
| 21:22.38 | marchon | does anyone know what I need - and where I might be able to get it shipped from today for delivery tomorrow? |
| 21:22.47 | rob0 | Add Tums to the list of hard drugs. |
| 21:22.51 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 21:23.01 | Innatech | Oh. It's going to be that kind of day. |
| 21:23.01 | [TK]D-Fender | marchon, What do you mean by "timing board" and why do you think you need this "board"? |
| 21:23.37 | [TK]D-Fender | rob0, They can be quite addictive... they're fruit flavoured and go down like Sweet Tarts :| |
| 21:23.50 | Innatech | I'll skip the obvious pun. |
| 21:23.55 | marchon | can not get ztdummy to compile and install . |
| 21:24.01 | holiday_42 | <snickers> |
| 21:24.09 | [TK]D-Fender | marchon, Perhaps we can help with that. |
| 21:24.19 | [TK]D-Fender | marchon, Why is it so far that you have failed? |
| 21:24.22 | *** join/#asterisk mrdigital (n=err@207-172-229-100.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 21:24.22 | marchon | and need to have menus - cepstral and such working |
| 21:24.33 | marchon | boss says - buy the right card - get it working tomorrow. |
| 21:24.44 | marchon | plus want meet me bridge. |
| 21:24.46 | Innatech | Last I checked, FedEx won' |
| 21:24.51 | Innatech | t deliver |
| 21:24.54 | Innatech | same day |
| 21:24.58 | [TK]D-Fender | marchon, ANY card will do, and ZTDUMMY works fine. |
| 21:25.00 | Innatech | wow. Time for a new keyboard. |
| 21:25.10 | marchon | so I need to know which is the least expensive board. |
| 21:25.18 | marchon | to meet the timing requirement. |
| 21:25.23 | [TK]D-Fender | Innatech, Mine is flakey too since a recent spill. Angle of attack on the keys is shot. |
| 21:25.35 | [TK]D-Fender | marchon, X100P would do the job. |
| 21:25.40 | marchon | excellent. |
| 21:25.41 | Innatech | mine seems to be firing random spacebar presses. Fun. |
| 21:25.42 | mrdigital | hey voicemeup? |
| 21:25.44 | [TK]D-Fender | marchon, 1 port TDM board too |
| 21:26.08 | marchon | and I can get those from digium? |
| 21:26.20 | cheshair | [TK]D-Fender: ok, i'll read the book. see you tomorrow and maaany thank for your help and patience |
| 21:26.25 | *** part/#asterisk cheshair (n=fabio@host-78-12-100-200.cust-adsl.tiscali.it) |
| 21:27.33 | Innatech | [TK] D-Fender : I've been liking the Enermax Aurora lately, since I can't seem to find decent enough 101 key generic beige boards anymore and can't use 500s near other humans. |
| 21:27.33 | marchon | thank you for the assistance... going to digium site. |
| 21:28.29 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 21:28.38 | Innatech | *can't use Model M's, that is. |
| 21:29.00 | *** join/#asterisk Foxygnu (n=FoX@2001:41d0:1:44c8:cafe:cafe:cafe:42) |
| 21:31.11 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 21:32.57 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 21:33.04 | *** join/#asterisk taupin974 (n=taupin97@89.237.79.244) |
| 21:34.23 | *** join/#asterisk cheshair (n=fabio@host-78-12-100-200.cust-adsl.tiscali.it) |
| 21:34.38 | kombi | sigh, switched back to 1.2, now I can hear her sey "you are currently the only person in the convo", but after that app_meetme seems to completely phase out.. |
| 21:34.47 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
| 21:35.16 | kombi | I'm about to eat you piano.. |
| 21:35.57 | lesouvage | I have put all the relevant info about my meetme and /dev/zap/pseudo that in't there problem on http://www.pastebin.be/3259. Any clue? |
| 21:37.18 | kombi | lesouvage: I just went through that, re-install everything in the order libpri, zaptel, *, addons, make sure zaptel is loaded when you make * |
| 21:38.36 | *** part/#asterisk cheshair (n=fabio@host-78-12-100-200.cust-adsl.tiscali.it) |
| 21:38.38 | *** join/#asterisk eatmypiano (n=eatmypia@host86-132-181-229.range86-132.btcentralplus.com) |
| 21:38.59 | *** join/#asterisk blepsoaf (n=pbaker@nnat-gw.adeptra.com) |
| 21:42.16 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 21:42.25 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 21:42.40 | xo8ox | guys one of our agents can't log off from her phone... what could be the problem ? |
| 21:42.54 | xo8ox | everytime we try to log off it keeps saying"that agent is already logged on" |
| 21:43.05 | Mercestes | xo8ox, mandatory overtime. |
| 21:43.16 | Mercestes | :D |
| 21:43.22 | xo8ox | whats that :) |
| 21:43.27 | xo8ox | what do i do |
| 21:44.22 | mosty | xo8ox, looks like the agent is trying to log on, not off. have you checked your dialplan? |
| 21:44.25 | xo8ox | and eventhough she is logged in she is not getting calls on her phone from the que |
| 21:44.57 | xo8ox | yesterday she could login and logout |
| 21:45.06 | xo8ox | but todeay she is logged in and can't log out |
| 21:45.24 | xo8ox | server config files have not been touched or changed unless they get changed on their own lol |
| 21:47.09 | mosty | what does the console/logs show? |
| 21:47.20 | [TK]D-Fender | xo8ox, Your description = useless. Pastebin the queue status befor the logout attempt and the CLI output of the logout attempt. |
| 21:47.31 | [TK]D-Fender | xo8ox, at verbose 10. |
| 21:50.39 | xo8ox | http://www.pastebin.ca/589267 |
| 21:50.41 | xo8ox | thats the CLI |
| 21:51.35 | NightMonkey | Howdy. I'm considering picking up an IAXy, but the Wikipedia article mentions that "a radio frequency choke must be applied to the telephone cable to prevent RFI." Is that true? |
| 21:51.55 | NightMonkey | Or did someone just get a lemon IAXy? |
| 21:52.31 | Qwell[] | NightMonkey: I think he's smoking something pretty good... |
| 21:52.34 | Qwell[] | or bad |
| 21:53.08 | mosty | maybe they had a really long/bad cable or something |
| 21:53.18 | NightMonkey | Time to put a "cite sources" on that sucker. ;) |
| 21:53.24 | Err | heh, there's no reason that RFI would be more or less of an issue with a particular interface, unless there's something *seriously* wrong with its design (like, for instance, it contains zero input filtering) |
| 21:53.35 | [TK]D-Fender | xo8ox, remove all parms from your agentcallbacklogin |
| 21:53.51 | Qwell[] | s/all parms from your // |
| 21:53.55 | NightMonkey | So, do any IAXy users here have an RFI choke on their IAXy? |
| 21:54.03 | xo8ox | where are those parms |
| 21:54.15 | Qwell[] | NightMonkey: I don't know what an RFI choke even IS, so no. |
| 21:55.03 | xo8ox | other agents can login and out with no problems |
| 21:55.05 | NightMonkey | Qwell[]: Looks like it's a RJ-11 male-male "joiner", from the image in the WP article. |
| 21:55.09 | xo8ox | only exten 422 has this issue |
| 21:55.13 | nohop | evening, ppls :) |
| 21:56.43 | Err | it's a filter that removes radio frequencies from the line (typically they're just inductor cores that you pass the wires through) |
| 21:57.09 | blepsoaf | Hello all, I'm having a strange issue with some conferencing ( meetme ). A few people have been complaining lately that conferencing has been fading in and out. We currently use polycom phones and a PRI t1 line. Is there somethign that I can look into that might be causing this issue.. FYI - the polycom phones are on their own separate switch. |
| 21:57.47 | *** join/#asterisk anthm_mobile (n=anthm@000-457-779.area4.spcsdns.net) |
| 21:57.51 | blepsoaf | Almost if someone is turning the volume up and down on some of the users |
| 21:58.46 | nohop | [TK]D-Fender: what's the advantage of the SPA-3102 over the 3000 ? |
| 21:59.25 | [TK]D-Fender | nohop, Can act as a router, stronger CPU, under WARRANTY. |
| 21:59.52 | [TK]D-Fender | nohop, Better physical design, more indicative lights, etc |
| 22:00.40 | nohop | hmm... okay.. |
| 22:01.40 | nohop | well i wouldn't really need the router functionality.. i have iptables for that... but stronger cpu? does that mean the codecs it uses are better aswell ? |
| 22:06.00 | rob0 | That reminds me of a related question. I have an SPA-2000. Seems like calls from one extension to the other are fine. Is compression not used on local (physical subnet) calls? |
| 22:06.50 | rob0 | I remember reading that it wouldn't work well with simultaneous calls on both FXS ports. |
| 22:06.55 | k31th | evening |
| 22:07.08 | k31th | any of you guys heard of a qsig trunk ? |
| 22:07.16 | mosty | rob0, how many g729 licences does the device have? |
| 22:07.25 | NightMonkey | Ack, can the IAXy not use FQDNs for configuration of Asterisk upstream servers? |
| 22:07.51 | *** join/#asterisk WindBack (n=Administ@host48.190-136-109.telecom.net.ar) |
| 22:08.33 | WindBack | What is the diference betwen asterisk 1.2 and 1.4 (is 1.4 unstable???) |
| 22:08.40 | WindBack | yet |
| 22:08.41 | rob0 | I didn't license g729. It only connects to the asterisk on the same subnet. |
| 22:10.21 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:11.03 | mosty | rob0, with ata's the two things that influence quality when doing simultaneous calls on the device at the codec in use (sometimes the ata only has one g729 licence) and the cpu power of the device |
| 22:11.11 | mosty | in my experience |
| 22:12.52 | nohop | [TK]D-Fender: but.. that 3000 thing is still pretty usable (cause that one's pretty easy to get here) |
| 22:13.29 | rob0 | I suppose I could just turn off compression ... asterisk handles everything from outside. |
| 22:13.30 | [TK]D-Fender | nohop, Why are you still considering the SPA-3000? |
| 22:13.54 | nohop | cause it's half the money ? :) |
| 22:14.20 | [TK]D-Fender | nohop, more power to you then. Go buy it and if you're ahppy, you're ahppy. |
| 22:14.33 | NightMonkey | ahippy? |
| 22:14.52 | nohop | hehe |
| 22:15.02 | [TK]D-Fender | nohop, I'm not going to spend any more time since you don't care if the 3102 may be superior or not or receive better firmwares,etc. Sipura is DEAD. |
| 22:15.06 | [TK]D-Fender | happy* |
| 22:15.14 | NightMonkey | ;) |
| 22:15.33 | *** join/#asterisk marchon (n=marchon@c-75-68-31-152.hsd1.nh.comcast.net) |
| 22:16.08 | nohop | [TK]D-Fender: nah, i dunno which one i'll buy... what would that extra processing power give me ? |
| 22:16.08 | marchon | does anyone know where I can get an X100P shipped today for delivery tomorrow? |
| 22:16.20 | marchon | the last hour has been unproductive. |
| 22:16.35 | [TK]D-Fender | nohop, T.38 support, perhaps enough CPU to run G.729 on BOTH interfaces, etc. |
| 22:16.57 | sandorp | any known problems with Polycom Soundpoint 301's reaching asterisk via NAT? I'm looking at buying a few online so I can't return them if I run into problems |
| 22:16.58 | [TK]D-Fender | nohop, ability to act as a router allowing for more flexible deployment. Warranty in case fo failure. |
| 22:17.14 | rob0 | marchon, it's after COB here, otherwise I'd send you one. :) |
| 22:17.17 | [TK]D-Fender | nohop, I'm not going to waste any more words on this. choose whatever you will. |
| 22:17.22 | nohop | warrenty is pretty nice indeed... :) |
| 22:17.31 | nohop | ok ok, i'll shut up about it :) |
| 22:17.45 | [TK]D-Fender | sandorp, 301 = OLD, and the IP 320 costs LESS and includes speakerphone, etc. |
| 22:18.03 | sandorp | but it has only 1 ethernet connection |
| 22:18.24 | [TK]D-Fender | sandorp, If you need a passthrough, IP-330 then |
| 22:18.31 | sandorp | thanks |
| 22:18.55 | mosty | marcan, they are no longer produced, and they're terrible quality. do you still want one? |
| 22:19.07 | mosty | marchon, that was for you |
| 22:19.23 | [TK]D-Fender | mosty, he just needs a hardware zaptel timing source. |
| 22:19.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:20.00 | mosty | oh, i know how you feel then |
| 22:20.20 | rob0 | Clones are still made, and can be had very cheaply. |
| 22:20.33 | sandorp | [TK]D-Fender: the Polycom 330 uses PoE ... do I need a special type of switch/router to power these things via the network cable? |
| 22:20.58 | [TK]D-Fender | sandorp, either that or buy a power brick for it,. |
| 22:21.15 | mosty | sandorp, yes you need a PoE switch, or an injector, and possibly a special cable |
| 22:21.25 | mosty | or what [TK]D-Fender said |
| 22:21.39 | [TK]D-Fender | no special cable.... |
| 22:22.24 | mosty | i've heard reports that some PoE equipment need special cables, i personally have always used regular cables without problems |
| 22:22.48 | sandorp | the ad I'm reading says they include a special cable |
| 22:23.04 | sandorp | but I will need the power adapter |
| 22:23.14 | [TK]D-Fender | mosty, the "special" cable for Polycoms was that the IP 30X/50X dod not NATIVELY support PoE, and their "special cable" has the negotiation" circuitry built in-line in a nugget. |
| 22:23.20 | [TK]D-Fender | speaking of which.... |
| 22:23.23 | [TK]D-Fender | telnet |
| 22:23.28 | [TK]D-Fender | .... |
| 22:23.42 | mosty | ahh, well there you go |
| 22:24.06 | mosty | like one of those injector kits then |
| 22:24.25 | [TK]D-Fender | sandorp, if I were you, FERGET about the power brick and buy a PoE injector. far more flexible in the long run |
| 22:24.36 | rob0 | Nugget says telnet is evil :) |
| 22:24.49 | Nugget | heh |
| 22:24.57 | [TK]D-Fender | mosty, Also the 320/330 takes a STANDARD power brick that you can buy seperately, NOT an injector. |
| 22:25.11 | [TK]D-Fender | Nugget - bot doesn't like me so.... |
| 22:25.24 | [TK]D-Fender | *thwap* |
| 22:25.29 | mosty | [TK]D-Fender, i have never seen a device that was purely PoE |
| 22:25.47 | [TK]D-Fender | mosty, You really don't try at all, do you ;) |
| 22:26.11 | [TK]D-Fender | mosty, Aastra 480i = easy sample |
| 22:26.55 | [TK]D-Fender | mosty, plenty more. basically IP phone use looks "corporate" and who wants bricks at every desk? Injectors are getting cheaper, and the phones themselves far more so. |
| 22:26.58 | mosty | i have more interesting hobbies than browsing power requirements of various devices, heh |
| 22:27.20 | mosty | everything i've seen so far has had the option of both |
| 22:27.33 | [TK]D-Fender | mosty, Me too, but thats a conversation for a more "intimate" environment ;) |
| 22:28.04 | [TK]D-Fender | mosty, However the power requirements of "devices" CAN play a role ;) |
| 22:31.39 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 22:31.41 | mosty | i am naive, ned ea teenager to explain that to me |
| 22:32.36 | *** join/#asterisk ReDNeQ- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 22:33.03 | *** join/#asterisk Corydon76-home (i=two@pdpc/supporter/sustaining/Corydon76-home) |
| 22:33.03 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 22:35.54 | *** join/#asterisk saftsack (n=saftsack@pD9E05EE2.dip.t-dialin.net) |
| 22:37.41 | awannabe | what package is fxotune part of? i cant find it on freebsd!! |
| 22:37.51 | *** join/#asterisk rdb_ (n=rdb@gw.avila.edu) |
| 22:38.40 | [TK]D-Fender | awannabe, good odds on ZAPTEL..... |
| 22:38.51 | awannabe | yeah its not though...intresting |
| 22:39.28 | sandorp | I believe I've had instances where installing from src did not install fxotune and friends |
| 22:39.41 | awannabe | thats what it looks like now |
| 22:40.01 | sandorp | just copy the exe's to /usr/sbin or something similar |
| 22:40.04 | awannabe | but i dont see anything about fxotune anywhere |
| 22:40.47 | sandorp | awannabe: nope, you won't get the man page either |
| 22:41.24 | awannabe | sandorp: did you just download zap sources and gmake them then, and just copied fxotune? |
| 22:41.36 | sandorp | awannabe: yes |
| 22:42.15 | awannabe | ahh ok, ill try that |
| 22:44.40 | *** join/#asterisk anthm_mobile (n=anthm@000-457-779.area4.spcsdns.net) |
| 22:44.43 | awannabe | fxotune has been removed complety from the zaptel-freebsd sources |
| 22:44.55 | awannabe | dang this echo cancelation! |
| 22:46.11 | *** join/#asterisk el_4_jinete (n=fabio@mail.pulxar.com.co) |
| 22:47.34 | *** part/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 22:48.10 | KDan | Is asterisk's Playback() function able to play back mp3s by default? |
| 22:48.17 | KDan | or do you have to do something special to get it to be able to do that? |
| 22:48.36 | sandorp | mp3 support must be compiled in |
| 22:48.47 | el_4_jinete | Hi, all |
| 22:49.14 | KDan | sandorp: ah, so the default debian package wouldn't have it? :-( |
| 22:49.18 | KDan | dang |
| 22:49.40 | el_4_jinete | I've a trouble |
| 22:49.48 | sandorp | KDan: just install libmp3 before attempting to compile asterisk |
| 22:49.58 | [TK]D-Fender | KDan, Playback can play any format there is a format_[type].so for. |
| 22:50.05 | [TK]D-Fender | sandorp, incorrect. |
| 22:50.17 | [TK]D-Fender | KDan, go download and install ASTERISk-ADDONS |
| 22:50.19 | el_4_jinete | Cli, shows the following error B-channel 0/23 successfully restarted on span 1, what is that? |
| 22:50.20 | sandorp | I'll shut up then |
| 22:50.22 | sandorp | :) |
| 22:50.57 | KDan | [TK]D-Fender: can't see it in the debian apt-cache search |
| 22:51.11 | [TK]D-Fender | KDan, * does not come with MP3 support by default because of LEGAL concerns |
| 22:51.15 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 22:51.30 | [TK]D-Fender | KDan, Screw packaged *. Download the source and compile it yourself |
| 22:52.00 | KDan | not if i can avoid it. last thing i need is to have to maintain dependencies and updates manually |
| 22:52.27 | [TK]D-Fender | KDan, *'s dependencies don't really change. |
| 22:52.29 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 22:52.47 | [TK]D-Fender | KDan, And get over it. the VAST majority have and are leading productive happy lives.... |
| 22:53.34 | KDan | if i have to i'll compile |
| 22:56.30 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 22:56.34 | kink0 | hello |
| 22:56.50 | kink0 | anyway to reduce bandwidth consuption ? to modify payload or packets ? |
| 22:56.51 | *** join/#asterisk Braxus (n=bhsieh@66.147.214.164) |
| 22:57.33 | [TK]D-Fender | kink0, Change codecs & protocols |
| 22:57.48 | kink0 | [TK]D-Fender, I need to use g729 and SIP |
| 22:58.17 | [TK]D-Fender | kink0, That's like saying "I can't change anything.... so how can I make it DIFFERENT?!?!" |
| 22:58.22 | kink0 | [TK]D-Fender, but average is about 28kbps per call, so 20 kbps is consumed in packets head, I think. |
| 22:58.44 | kink0 | [TK]D-Fender, no exactly, I read about the consumption due to header |
| 22:58.59 | kink0 | but I did not found about how to compress RTP headers in Asterisk |
| 22:59.17 | [TK]D-Fender | kink0, RTP isn't the problem... UDP is. |
| 22:59.45 | [TK]D-Fender | kink0, see if you can change the frame size. |
| 22:59.48 | kink0 | yes, UDP |
| 23:00.04 | kink0 | that is what I pretend, but I Did not found the way |
| 23:00.21 | [TK]D-Fender | kink0, by default it uses 20ms packets IIRC. Keep in mind cetain endpoint might not be able to adaptt o your change |
| 23:00.28 | kink0 | I used the web bandwidth calculator for asterisk, to try differents values of sizes |
| 23:00.56 | kink0 | is not auto-adaptative if the origination uses a different payload ? |
| 23:01.22 | el_4_jinete | Any person that help's me? B-channel 0/23 successfully restarted on span 1, what is that? |
| 23:01.41 | kink0 | el_4_jinete, eso es normal, ni caso. |
| 23:03.02 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 23:03.11 | kink0 | [TK]D-Fender, ussually you requires to optimize QoS with tc and so ? |
| 23:03.12 | [TK]D-Fender | kink0, No. |
| 23:03.25 | kink0 | ( I have prioritized UDP in the cisco gateway router ) |
| 23:03.26 | [TK]D-Fender | kink0, QoS has NOTHING to do with anything. |
| 23:03.48 | [TK]D-Fender | kink0, Change your voice frame size. Thats IT. PERIOD. |
| 23:03.59 | JT | el_4_jinete: that's normal |
| 23:04.01 | [TK]D-Fender | kink0, And hope that you other devices can comply |
| 23:04.19 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 23:04.33 | kink0 | [TK]D-Fender, now the newbie question... where to modify the size ? ip or asterisk ? |
| 23:04.39 | el_4_jinete | JT, but when that happend, all the calls were down |
| 23:04.42 | [TK]D-Fender | el_4_jinete, There is a "resetinterval" parm you can set to prevent the regular resets that * causes |
| 23:04.49 | [TK]D-Fender | kink0, In the codec. |
| 23:05.09 | kink0 | el_4_jinete, then your T1/E1 is wrongs cofigured, may be you got any claim about D-channel ? |
| 23:06.06 | kink0 | [TK]D-Fender, hmmmm... not codec.conf ? g729.h and recompile ? I am mainly ussing the g729 from Digium, due to license , but I have tried also the Intel IPP |
| 23:06.44 | el_4_jinete | kink0, I review the settings any times, and the D-Channel shows HDLS bad FCS |
| 23:07.05 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 23:07.21 | kink0 | el_4_jinete, the reset of one channel, must not drop calls on all channels, if that happens, you have a mistake in your zaptel.conf or zapata.conf |
| 23:07.33 | [TK]D-Fender | gah |
| 23:07.41 | kink0 | el_4_jinete, T1/E1 ? what country ? |
| 23:07.47 | el_4_jinete | [TK]D-Fender, what's that parm? |
| 23:07.55 | KDan | There you go |
| 23:08.10 | KDan | http://pkg-voip.buildserver.net/debian/ |
| 23:08.15 | KDan | Has the asterisk-mp3 package |
| 23:08.19 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:08.22 | [TK]D-Fender | el_4_jinete, I just TOLD YOU. |
| 23:08.24 | [TK]D-Fender | resetinterval: sets the time in seconds between restart of unused channels, defaults to |
| 23:08.24 | [TK]D-Fender | 3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set the interval to a |
| 23:08.24 | [TK]D-Fender | very long interval e.g. 100000000 or 'never' to disable *entirely*. |
| 23:08.30 | el_4_jinete | kink0, I'm in Colombia, the signalling is euroisdn |
| 23:08.53 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 23:08.53 | kink0 | [TK]D-Fender, appears all channels are droped when one is reseted |
| 23:08.55 | el_4_jinete | [TK]D-Fender, sorry... I see |
| 23:08.58 | sci_05 | are there jitter setting for sip? |
| 23:09.21 | kink0 | el_4_jinete, hmmm I am ussing also euroisdn, I can pass you my zaptel and zapata files , so you can compare with yours |
| 23:09.29 | el_4_jinete | [TK]D-Fender, Yes all channels, and all calls were down |
| 23:09.31 | kink0 | I have never experienced that problem |
| 23:09.42 | kink0 | well, just if I lost the D, of course |
| 23:09.47 | [TK]D-Fender | el_4_jinete, set to "never" then |
| 23:09.57 | el_4_jinete | kink0, OK |
| 23:10.53 | anonymouz666 | I have Mysql() app doing a SQL count. and that returns a value. let's say 3. then I need to populate this number (3) into another function parameters... func(par1, par2, par3). Anyone has an idea how can I do that through dialplan? |
| 23:11.09 | *** join/#asterisk anthm (n=anthm@dhcp64-134-34-252.bwic.chi.wayport.net) |
| 23:11.09 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:12.15 | [TK]D-Fender | anonymouz666, what does it return it INTO? |
| 23:12.19 | el_4_jinete | Bueno, pues yo lo tengo asi |
| 23:12.28 | el_4_jinete | span=1,1,0,ccs,hdb3 |
| 23:12.28 | el_4_jinete | bchan=1-15 |
| 23:12.28 | el_4_jinete | dchan=16 |
| 23:12.30 | el_4_jinete | bchan=17-31 |
| 23:12.38 | el_4_jinete | loadzone=fr |
| 23:12.39 | el_4_jinete | defaultzone=fr |
| 23:13.36 | kink0 | el_4_jinete, ojo al CRC , si tu proveedor lo tiene activado, tu necesitas tambien tenerlo |
| 23:14.05 | anonymouz666 | [TK]D-Fender: if count() return 3 lines. I need to have 3 vars to store 3 differents values |
| 23:14.18 | JT | ~pb |
| 23:14.19 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 23:14.23 | JT | el_4_jinete: use pastebin |
| 23:14.33 | el_4_jinete | JT, OK |
| 23:15.10 | [TK]D-Fender | anonymouz666, ok... and? |
| 23:15.16 | *** join/#asterisk wag3slav3 (n=gregg@71-32-119-21.bsmr.qwest.net) |
| 23:15.23 | el_4_jinete | kink0, pues mira. Yo tenia el crc4, pero llamé al provedor y le dije que lo quitara. Y tengo el mismo problema, es más, hasta cambie la TE110p, por otra |
| 23:15.38 | el_4_jinete | Y nada... :( |
| 23:16.00 | wag3slav3 | I am having trouble getting asterisk to detect an 800 number that auto-answers with no ring. Can anyone help with answer detection? |
| 23:16.26 | *** join/#asterisk lymeca (n=lymeca@unaffiliated/lymeca) |
| 23:16.42 | [TK]D-Fender | wag3slav3, numbers don't ring, DEVICES DO. Same goes for ASNWERING. |
| 23:16.51 | [TK]D-Fender | wag3slav3, DETAILS might help. |
| 23:16.58 | el_4_jinete | kink0, ademas mi zapata.conf esta igualito al tuyo. |
| 23:18.01 | wag3slav3 | Outbound sip call to an 800 number. Whatever device is at the other end auto answers. My asterisk doesn't detect the active channel because it never gets a ring status. various sip devices on my end. |
| 23:18.02 | kink0 | el_4_jinete, see about the PCI clocking, try to fix it to 66Mhz instead auto, I discover some boards problems with over 66Mhz PCI clocking |
| 23:18.38 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 23:18.57 | [TK]D-Fender | wag3slav3, So * never thinks the call is actually connected? |
| 23:19.07 | wag3slav3 | That's right. |
| 23:19.09 | wag3slav3 | It just times out. |
| 23:19.11 | Daejeo1 | TK]D-Fender:yo |
| 23:19.18 | [TK]D-Fender | wag3slav3, Sounds like your ITSP is BS. Pick a new one. |
| 23:19.24 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-104-68-39.pskn.east.verizon.net) |
| 23:19.29 | wag3slav3 | PSTN T1. |
| 23:19.30 | Daejeo1 | :) |
| 23:19.36 | el_4_jinete | kink0, Listo yo lo hago. Por ahora modifique en zapata.conf resetinterval=never. Tomorrow I will tell us. |
| 23:19.41 | el_4_jinete | Thanks all |
| 23:19.43 | [TK]D-Fender | wag3slav3, that does NOT add up. |
| 23:20.01 | wag3slav3 | Fender: I hear you. |
| 23:20.08 | Daejeo1 | TK]D-Fender: What does one computer say to an other computer?? |
| 23:20.13 | [TK]D-Fender | wag3slav3, What does "PSTN T1" have to do with "SIP ITSP"? |
| 23:20.27 | [TK]D-Fender | Daejeo1, Nothing... they just shake hands ;) |
| 23:20.36 | Daejeo1 | no |
| 23:20.38 | wag3slav3 | Fender: Well, my end is a sip device, routed out on a PSTN T1, therefore I do no have an ITSP. |
| 23:20.39 | Daejeo1 | 0101010 |
| 23:20.53 | Qwell[] | 7 bit bytes? |
| 23:20.57 | Daejeo1 | say 01010101 |
| 23:21.10 | [TK]D-Fender | wag3slav3, You just said "sip call to 800 number". What the hell am I supposed to think about that? |
| 23:21.21 | [TK]D-Fender | wag3slav3, Sure SOUNDS like an ITSP to me. |
| 23:21.28 | [TK]D-Fender | wag3slav3, you need to be more clear. |
| 23:21.51 | [TK]D-Fender | wag3slav3, Now start over and be PRECISE in your description of EXACTLY what hardware is involved. |
| 23:22.32 | [TK]D-Fender | Qwell[], Serial: E71 ;) |
| 23:23.55 | wag3slav3 | Fender: Sorry, my device is a sip device, to asterisk, then my T1. Then the other end doesn't ring. I have tried it with my ITSP backup trunk, I get the same error. I have a grandstream 496 routed via sip to asterisk, trunks out via a PSTN T1 hooked to a Audiocodes Mediant 1000 to this destination. |
| 23:24.00 | wag3slav3 | Sorry about the vaugness. |
| 23:24.07 | alrs | Innatech: back to the T1 stuff, it looks like AM6 are big in to metro ethernet |
| 23:24.19 | alrs | Innatech: which would explain why their T1 only does data |
| 23:24.31 | [TK]D-Fender | wag3slav3, So your MEDIANT is not reporting "answer" back to *? |
| 23:24.52 | wag3slav3 | Fender: In my testing I have tried several different sip devices, as well as to my backup ITSP, which also doens't report answer back. |
| 23:25.25 | [TK]D-Fender | wag3slav3, Stop being generic about the tech people are using! |
| 23:25.26 | *** part/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 23:25.59 | JT | wag3slav3: my devices and widgets don't work, help! |
| 23:26.03 | *** join/#asterisk dansmith (n=dan@gw0.danplanet.com) |
| 23:26.11 | [TK]D-Fender | wag3slav3, Mediant 1000 is not an ITSP, it is a SIP gateway. And what is this ITSP (you said there WAS NONE before, now you're changing things...) using / doing? |
| 23:26.45 | Daejeo1 | JT :) |
| 23:26.55 | [TK]D-Fender | wag3slav3, enable SIP debug, place a call and SHOW US what the heck is going on. in a PASTEBIN. |
| 23:26.57 | [TK]D-Fender | ~pb |
| 23:26.57 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 23:26.59 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 23:27.07 | wag3slav3 | thank you. |
| 23:27.25 | anonymouz666 | [TK]D-Fender: the logic is simple. count returns 3 lines. then I have to store 3 values with MySQL fetchid ${var1} ${var2} ${var3}. for each var1 I will have an gotoif condition. |
| 23:27.32 | anonymouz666 | just that |
| 23:28.17 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 23:28.26 | [TK]D-Fender | anonymouz666, APSTEBIN.... |
| 23:28.29 | [TK]D-Fender | PASTEBIN* |
| 23:29.04 | anonymouz666 | ok |
| 23:29.16 | [TK]D-Fender | anonymouz666, prototyepe some code and we'll see what we can do |
| 23:29.28 | anonymouz666 | right |
| 23:33.28 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 23:34.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:35.51 | *** join/#asterisk dlynes_laptop (n=dlynes@216.19.191.152.novuscom.net) |
| 23:38.03 | *** join/#asterisk SwK (n=SwK@206.166.206.34) |
| 23:38.12 | JT | look like pb was too hard for wag3slav3 |
| 23:39.59 | [TK]D-Fender | JT : Feel free to count the tears I am NOT crying... |
| 23:40.10 | JT | heh |
| 23:44.07 | *** join/#asterisk mihinomenest (i=m0B9@cerebus.clandestineresearch.com) |
| 23:44.53 | anonymouz666 | [TK]D-Fender: http://pastebin.ca/589403 |
| 23:44.59 | mihinomenest | how much of a red-herring is variable-length DTMF? |
| 23:45.11 | darius_ | Is there a tool that can take a sip and/or iax call packet trace and extract the audio from it? |
| 23:45.23 | JT | yes |
| 23:45.55 | darius_ | and what would that tool be? :) |
| 23:46.07 | JT | wireshark |
| 23:46.12 | Qwell[] | pretty sure wireshark can do sip/rtp |
| 23:46.16 | Qwell[] | not sure about iax2 though |
| 23:46.30 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 23:46.33 | darius_ | but it can extract the audio - to a file? |
| 23:46.51 | JT | yes |
| 23:46.55 | JT | on sip/rtp |
| 23:47.40 | darius_ | huh. |
| 23:47.45 | darius_ | what would be the option to do that? |
| 23:48.05 | [TK]D-Fender | anonymouz666, what is the COUNT returned in |
| 23:48.09 | JT | i dunno, i imagine google could help |
| 23:48.52 | *** join/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net) |
| 23:49.16 | darius_ | fair enough |
| 23:50.12 | anonymouz666 | [TK]D-Fender: the time conditions... a user may have one or more conditions |
| 23:50.43 | [TK]D-Fender | anonymouz666, Wrong answer. WHERE is the COUNT returned? |
| 23:51.35 | [TK]D-Fender | anonymouz666, and I am already rather certain that you have several conditions, but I'm not licensed to practice currently. |
| 23:52.15 | *** join/#asterisk SwK (n=SwK@206.166.206.34) |
| 23:55.19 | lymeca | I'm trying to gain a basic understanding of how Asterisk can benefit me as an SIP softphone (currently Ekiga) user. I'm used to just connecting directly to another peer using SIP (hopefully with a ZRTP enabled client) to talk with them. |
| 23:55.45 | anonymouz666 | [TK]D-Fender: exten => 10,n,Mysql(Fetch fetchid ${resultid} ${cond_loop}) |
| 23:55.54 | anonymouz666 | I forgot to include this line |
| 23:56.08 | lymeca | I could put Asterisk on my server with a static global IP address, but how could that change my SIP usage for the better? |
| 23:56.13 | anonymouz666 | ${cond_loop} has the number of the lines |
| 23:56.15 | *** join/#asterisk dlucas (n=root@Broadband-Dynamic-Western120.connect.com.fj) |
| 23:56.19 | anonymouz666 | 3 or 4 or 5... |
| 23:56.23 | dlucas | hi |
| 23:56.24 | JT | which'd be great if asterisk could do ZRTP |
| 23:56.41 | dlucas | dlucas |
| 23:56.49 | dlucas | sorry |
| 23:57.39 | [TK]D-Fender | anonymouz666, So what do you do with each record retreived? Add the line in, give a "1" priority somewhere..... |
| 23:57.41 | dlucas | wondering if someone is able to explain why I get the followin error |
| 23:57.45 | dlucas | Internal RTCP NTP clock skew detected: |
| 23:58.34 | [TK]D-Fender | lymeca, if all you want to do is talk with someone over a VoIP protocal and they have a client like Ekiga you don't NEED *. |
| 23:58.37 | anonymouz666 | each record retreived are stored into a var. (line 3) |
| 23:58.49 | [TK]D-Fender | anonymouz666, New PB please. |
| 23:58.54 | anonymouz666 | I am added manually 2 vars. because I know the count will return that |
| 23:59.09 | dlucas | n |
| 23:59.11 | anonymouz666 | exten => 10,n,Mysql(Fetch fetchid ${resultid} condicaotempo app-condicaotempo) |
| 23:59.13 | anonymouz666 | this is line 3 |
| 23:59.14 | Qwell[] | Where is Syracuse? |
| 23:59.16 | kombi | dlucas: I had the same thing about 2 hours ago, downgraded to 1.2 and it is gone but other stuff still isn't working |
| 23:59.19 | Qwell[] | (my random question of the day) |
| 23:59.34 | [TK]D-Fender | anonymouz666, I'm not going to repeat myself. |
| 23:59.41 | *** join/#asterisk coppice (n=chatzill@163.201.17.210.dyn.pacific.net.hk) |
| 23:59.48 | anonymouz666 | oh nevermind. |
| 23:59.50 | anonymouz666 | forget it. |
| 23:59.52 | dlucas | the error message has only been introduced in 1.4.5 |
| 23:59.56 | dlucas | 1.4.4 did not have it |
| 23:59.59 | kombi | Qwell: you mean the one in Sicilly.. |